1 ==============================================================================
3 === THIS FILE IS AUTOMATICALLY GENERATED DURING THE RELEASE
4 === PROCESS. DO NOT MAKE CHANGES HERE. INSTEAD, REFER TO
5 === doc/CHANGES-staging/README.md FOR MORE DETAILS.
7 === This file documents the new and/or enhanced functionality added in
8 === the Asterisk versions listed below. This file does NOT include
9 === changes in behavior that would not be backwards compatible with
10 === previous versions; for that information see the UPGRADE.txt file
11 === and the other UPGRADE files for older releases.
13 ==============================================================================
15 ------------------------------------------------------------------------------
16 --- New functionality introduced in Asterisk 18.0.0 --------------------------
17 ------------------------------------------------------------------------------
21 * The Streams API becomes the home for the core ACN capabilities.
24 * Parsing and formatting of codec negotation preferences.
25 * Resolving pending streams and topologies with those configured
26 using configured preferences.
27 * Utility functions for creating string representations of
28 streams, topologies, and negotiation preferences.
30 For codec negotiation preferences:
31 * Added ast_stream_codec_prefs_parse() which takes a string
32 representation of codec negotiation preferences, which
33 may come from a pjsip endpoint for example, and populates
34 a ast_stream_codec_negotiation_prefs structure.
35 * Added ast_stream_codec_prefs_to_str() which does the reverse.
36 * Added many functions to parse individual parameter name
37 and value strings to their respectrive enum values, and the
41 * Added ast_stream_create_resolved() which takes a "live" stream
42 and resolves it with a configured stream and the negotiation
43 preferences to create a new stream.
44 * Added ast_stream_to_str() which create a string representation
45 of a stream suitable for debug or display purposes.
48 * Added ast_stream_topology_create_resolved() which takes a "live"
49 topology and resolves it, stream by stream, with a configured
50 topology stream and the negotiation preferences to create a new
52 * Added ast_stream_topology_to_str() which create a string
53 representation of a topology suitable for debug or display
55 * Renamed ast_format_caps_from_topology() to
56 ast_stream_topology_get_formats() to be more consistent with
57 the existing ast_stream_get_formats().
60 * A new function ast_format_cap_append_names() appends the results
61 to the ast_str buffer instead of replacing buffer contents.
65 * The BridgeAdd application now behaves more like the Bridge application.
66 The application now sets the BRIDGERESULT channel variable to indicate
67 what happened when the channel resumes in dialplan. This is instead of
68 hanging up the channel on failure conditions.
72 * Two new options, incoming_call_offer_pref and outgoing_call_offer_pref
73 have been added to res_pjsip endpoints that specify the preferred order
74 of codecs to use between those received/sent in an SDP offer and those
75 set in the endpoint configuration.
77 ------------------------------------------------------------------------------
78 --- Functionality changes from Asterisk 17.0.0 to Asterisk 18.0.0 ------------
79 ------------------------------------------------------------------------------
83 * You can now specify an optional 'Content-Type' as an argument for the Asterisk
84 SendText manager action.
88 * A new parameter 'inhibitConnectedLineUpdates' is now available in the
89 'bridges.addChannel' call. This prevents the identity of the newly connected
90 channel from being presented to other bridge members.
94 * The Channel resource has a new sub-resource "externalMedia".
95 This allows an application to create a channel for the sole purpose
96 of exchanging media with an external server. Once created, this
97 channel could be placed into a bridge with existing channels to
98 allow the external server to inject audio into the bridge or
99 receive audio from the bridge.
100 See https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI
101 for more information.
105 * H.265/HEVC is now a supported video codec and it can be used by
106 specifying "h265" in the allow line.
107 Please note however, that handling of the additional SDP parameters
108 described in RFC 7798 section 7.2 is not yet supported.
112 * Adds support for AudioSocket, a very simple bidirectional audio streaming
113 protocol. There are both channel and application interfaces.
115 A description of the protocol can be found on the referenced wiki page. A
116 short talk about the reasons and implementation can be found on YouTube at
119 ARI support has also been added via the existing "externalMedia" ARI
120 functionality. The UUID is specified using the arbitrary "data" field.
122 Wiki: https://wiki.asterisk.org/wiki/display/AST/AudioSocket
123 YouTube: https://www.youtube.com/watch?v=tjduXbZZEgI
127 * In order to reduce the amount of AMI and ARI events generated,
128 the global "Message/ast_msg_queue" channel can be set to suppress
129 it's normal channel housekeeping events such as "Newexten",
130 "VarSet", etc. This can greatly reduce load on the manager
131 and ARI applications when the Digium Phone Module for Asterisk
132 is in use. To enable, set "hide_messaging_ami_events" in
133 asterisk.conf to "yes" In Asterisk versions <18, the default
134 is "no" preserving existing behavior. Beginning with
135 Asterisk 18, the option will default to "yes".
139 * STIR/SHAKEN support has been added to Asterisk. Configuration is done in
140 stir_shaken.conf. There is a sample configuration file to help you get
141 started (asterisk/configs/samples/stir_shaken.conf.sample). Once that's
142 set up, you can enable STIR/SHAKEN on any endpoint by setting stir_shaken
143 to yes on the endpoint configuration object. This will add an Identity
144 header on outgoing INVITEs, and check for an Identity header on incoming
145 INVITEs. This option has been added to Alembic as well.
147 The information received on an incoming INVITE can be checked using the
148 STIR_SHAKEN dialplan function. There are two variations:
151 STIR_SHAKEN(0, verify_result)
153 The first variation will tell you how many STIR/SHAKEN results are on the
154 channel. The second fetches information for a specific result. The first
155 parameter is the index, followed by what information you want to retrieve.
156 The available options are 'verify_result', 'identity', and 'attestation'.
160 * The ChanIsAvail application now tolerates empty positions in the supplied
161 device list. Dialplan can now be simplified by not having to check for
162 empty positions in the device list.
166 * A new bridge profile option, maximum_sample_rate, has been added which sets
167 a maximum sample rate that the bridge will be mixed at. This allows the bridge
168 to move below the maximum sample rate as needed but caps it at the maximum.
170 * A new option, "text_messaging", has been added to the user profile
171 which allows control over whether text messaging is enabled or
172 disabled for a user. If enabled (the default) text messages
173 will be sent to the user. If disabled no text messages will be
178 * The Dial application now tolerates empty positions in the supplied
179 destination list. Dialplan can now be simplified by not having to check
180 for empty positions in the destination list. If there are no endpoints to
181 dial then DIALSTATUS is set to CHANUNAVAIL.
185 * An option 'S' has been added to MixMonitor. If used in combination with
186 the r() and/or t() options, if a frame is available to write to one of
187 those files but not the other, a frame of silence if written to the file
188 that does not have an audio frame. This should prevent the two files
189 from "drifting" when mixed after the fact.
191 * If the 'filename' argument to MixMonitor() ended with '.wav49,'
192 Asterisk would silently convert the extension to '.WAV' when opening
193 the file for writing. This caused the MIXMONITOR_FILENAME variable to
194 reference the wrong file. The MIXMONITOR_FILENAME variable will now
195 reflect the name of the file that Asterisk actually used instead of
196 the filename that was passed to the application.
200 * The Page application now tolerates empty positions in the supplied
201 destination list. Dialplan can now be simplified by not having to check
202 for empty positions in the destination list.
206 * A feature was added in Asterisk 13.27.0 and 16.4.0 that removed lock files from
207 the Asterisk voicemail directory on startup. Some users that store their
208 voicemails on network storage devices experienced slow startup times due to the
209 relative expense of traversing the voicemail directory structure looking for
210 orphaned lock files. This feature has now been removed.
212 Users who require the lock files to be removed at startup should modify their
213 startup scripts to do so before starting the asterisk process.
217 * A new dialplan function, PJSIP_MOH_PASSTRHOUGH, has been added to chan_pjsip. This
218 allows the behaviour of the moh_passthrough endpoint option to be read or changed
219 in the dialplan. This allows control on a per-call basis.
223 * The UnicastRTP channel driver provided by chan_rtp now accepts
224 "<hostname>:<port>" as an alternative to "<ip_address>:<port>" in the destination.
225 The first AAAA (preferred) or A record resolved will be used as the destination.
226 The lookup is synchronous so beware of possible dialplan delays if you specify a
231 * A new parameter, httpheader, has been added to CURLOPT function. This parameter
232 allows to set custom http headers for subsequent calls off CURL function.
233 Any setting of headers will replace the default curl headers
234 (e.g. "Content-type: application/x-www-form-urlencoded")
236 * A new option, followlocation, can now be enabled with the CURLOPT()
237 dialplan function. Setting this will instruct cURL to follow 3xx
238 redirects, which it does not by default.
242 * The JITTERBUFFER dialplan function now has an option to enable video synchronization
243 support. When enabled and used with a compatible channel driver (chan_sip, chan_pjsip)
244 the video is buffered according to the size of the audio jitterbuffer and is
245 synchronized to the audio.
249 * Accept decimal number as argument.
253 * You can now disable the /httpstatus page served by Asterisk's built-in
254 HTTP server by setting 'enable_status' to 'no' in http.conf.
258 * The 'minmemfree' configuration option now counts memory allocated to
259 the filesystem cache as "free" because it is memory that is available
264 * When creating a channel in ARI using the create call
265 you can now specify dialplan variables to be set as part
266 of the same operation.
270 * This fix allows a realtime moh class to be unregistered from the command
271 line. This is useful when the contents of a directory referenced by a
272 realtime moh class have changed.
273 The realtime moh class is then reloaded on the next request and uses the
274 new directory contents.
276 * A new mode - playlist - has been added to res_musiconhold. This mode allows the
277 user to specify the files (or URLs) to play explicitly by putting them directly
282 * Added a new PJSIP system setting called disable_rport.
283 Default is no to keep support working as before.
285 If it is false (default) it adds the 'rport' parameter in the outgoing request message.
286 If it is true it does not add the 'rport' parameter in the outgoing request message.
288 This is a system option, but working as a global option.
290 res_pjsip_endpoint_identifier_ip
292 * In 'type = identify' sections, the addresses specified for the 'match'
293 clause can now include a port number. For IP addresses, the port is
294 provided by including a colon after the address, followed by the
295 desired port number. If supplied, the netmask should follow the port
296 number. To specify a port for IPv6 addresses, the address itself must
297 be enclosed in brackets to be parsed correctly.
301 * The PJSIP packet logger now has the following CLI commands:
303 pjsip set logger pcap <filename>
305 When used this will create a pcap file containing the incoming
306 and outgoing SIP packets, in unencrypted form.
308 pjsip set logger console <on / off>
310 This allows you to toggle logging to console on and off.
312 pjsip set logger host <IP/subnet mask> add
314 This allows you to add an additional IP address or subnet
315 mask to logging, allowing you to log multiple instead of
316 just a single IP address or all traffic.
318 The normal "pjsip set logger host" CLI command has also been
319 expanded to allow subnet masks as well.
323 * When placing an outgoing call to a PJSIP endpoint the intent
324 of any requested formats will now be respected. If only an audio
325 format is requested (such as ulaw) but the underlying endpoint
326 does not support the format the resulting SDP will still only
327 contain an audio stream, and not any additional streams such as
330 * Two new options, incoming_call_offer_pref and outgoing_call_offer_pref
331 have been added to res_pjsip endpoints that specify the preferred order
332 of codecs to use between those received/sent in an SDP offer and those
333 set in the endpoint configuration.
337 * This change include a new cli command 'rtp show settings'
339 The command display by general settings of rtp configuration. For this
340 point is added the fields: rtpstart, rtpend, dtmftimeout, rtpchecksum,
341 strictrtp, learning_min_sequential and icesupport.
343 * The blacklist mechanism in res_rtp_asterisk for ICE and STUN was converted to
346 As such six now options are now available:
355 These options have their obvious meanings as used elsewhere.
357 Backwards compatibility was maintained by adding {stun,ice}_blacklist as
358 aliases for {stun,ice}_deny.
360 res_sorcery_memory_cache
362 * The SorceryMemoryCacheExpireObject AMI action and CLI
363 command allow expiring of a specific object within the
364 sorcery memory cache. This is done by removing the
365 object from the cache with the expectation that the
366 cache will then re-populate the object when it is next
369 For full backend caching this does not occur. The cache
370 won't repopulate until an entire refresh is done resulting
371 in the possibility that objects are missing until that
374 The AMI action and CLI command will now not allow
375 expiring of an object if the cache is configured as a
376 full backend cache. Instead you must use either the
377 SorceryMemoryCacheExpire or SorceryMemoryCachePopulate
378 AMI actions or their associated CLI commands.
382 * Added two new CLI commands to reset stats for taskprocessors. You can
383 reset stats for a single, specific taskprocessor ('core reset
384 taskprocessor <taskprocessor>'), or you can reset all taskprocessors
385 ('core reset taskprocessors'). These commands will reset the counter for
386 the number of tasks processed as well as the max queue size.
388 * Added "like" support for 'core show taskprocessors'. Now you
389 can specify a specific set of taskprocessors (or just one) by
390 adding the keyword "like" to the above command, followed by
391 your search criteria.
393 ------------------------------------------------------------------------------
394 --- New functionality introduced in Asterisk 17.0.0 --------------------------
395 ------------------------------------------------------------------------------
399 * The bridging core no longer uses the stasis cache for bridge
400 snapshots. The latest bridge snapshot is now stored on the
401 ast_bridge structure itself.
403 The following APIs are no longer available since the stasis cache
405 ast_bridge_topic_cached()
406 ast_bridge_topic_all_cached()
408 A topic pool is now used for individual bridge topics.
410 The ast_bridge_cache() function was removed since there's no
411 longer a separate container of snapshots.
413 A new function "ast_bridges()" was created to retrieve the
414 container of all bridges. Users formerly calling
415 ast_bridge_cache() can use the new function to iterate over
416 bridges and retrieve the latest snapshot directly from the
419 The ast_bridge_snapshot_get_latest() function was renamed to
420 ast_bridge_get_snapshot_by_uniqueid().
422 A new function "ast_bridge_get_snapshot()" was created to retrieve
423 the bridge snapshot directly from the bridge structure.
425 The ast_bridge_topic_all() function now returns a normal topic
426 not a cached one so you can't use stasis cache functions on it
429 The ast_bridge_snapshot_type() stasis message now has the
430 ast_bridge_snapshot_update structure as it's data. It contains
431 the last snapshot and the new one.
435 * The core no longer uses the stasis cache for channels snapshots.
436 The following APIs are no longer available:
437 ast_channel_topic_cached()
438 ast_channel_topic_all_cached()
439 The ast_channel_cache_all() and ast_channel_cache_by_name() functions
440 now returns an ao2_container of ast_channel_snapshots rather than a
441 container of stasis_messages therefore you can't call stasis_cache
443 The ast_channel_topic_all() function now returns a normal topic,
444 not a cached one so you can't use stasis cache functions on it either.
445 The ast_channel_snapshot_type() stasis message now has the
446 ast_channel_snapshot_update structure as it's data.
447 ast_channel_snapshot_get_latest() still returns the latest snapshot.
451 * The chan_sip module is now deprecated, users should migrate to the
452 replacement module chan_pjsip. See guides at the Asterisk Wiki:
453 https://wiki.asterisk.org/wiki/x/tAHOAQ
454 https://wiki.asterisk.org/wiki/x/hYCLAQ
456 ------------------------------------------------------------------------------
457 --- Functionality changes from Asterisk 16.0.0 to Asterisk 17.0.0 ------------
458 ------------------------------------------------------------------------------
462 * A new application, this will queue up attended transfer to the given extension.
466 * A new application, this will redirect all channels currently
467 bridged to the caller channel to the specified destination.
471 * Add "average_all", "highest_all", and "lowest_all" values for
472 the remb_behavior option. These values operate on a bridge
473 level instead of a per-source level. This means that a single
474 REMB value is calculated and sent to every sender, instead of
475 a REMB value that is unique for the specific sender..
479 * Add RINGTIME and RINGTIME_MS variables containing respectively seconds and
480 milliseconds between creation of the dialing channel and receiving the first
483 Add PROGRESSTIME and PROGRESSTIME_MS variables analogous to the above with respect to
484 the PROGRESS signal. Shorter of these two times should be equivalent to
485 the PDD (Post Dial Delay) value
487 Add DIALEDTIME_MS and ANSWEREDTIME_MS variables to get millisecond resolution
488 versions of DIALEDTIME and ANSWEREDTIME
492 * You can now indicate that you'd like an ice_host_candidate's local address
493 to be published as well as the mapped address. See the sample rtp.conf
494 for more information.
498 * Add 'p' option to stop reading extension if user presses '#' key.
502 * The DUNDi PBX module now supports IPv4/IPv6 dual binding.
506 * Added a new PJSIP global setting called norefersub.
507 Default is true to keep support working as before.
509 res_pjsip_refer configures PJSIP norefersub capability accordingly.
511 Checks the PJSIP global setting value.
512 If it is true (default) it adds the norefersub capability to PJSIP.
513 If it is false (disabled) it does not add the norefersub capability
516 This is useful for Cisco switches that do not follow RFC4488.
520 * DTLS packets will now be fragmented according to the MTU as set in rtp.conf. This
521 allows larger certificates to be used for the DTLS negotiation. By default this value
524 ------------------------------------------------------------------------------
525 --- Functionality changes from Asterisk 16.2.0 to Asterisk 16.3.0 ----------
526 ------------------------------------------------------------------------------
530 * Application event filtering is now supported. An application can now specify
531 an "allowed" and/or "disallowed" list(s) of event types. Only those types
532 indicated in the "allowed" list are sent to the application. Conversely, any
533 types defined in the "disallowed" list are not sent to the application. Note
534 that if a type is specified in both lists "disallowed" takes precedence.
536 * A new REST API call has been added: 'move'. It follows the format
537 'channels/{channelId}/move' and can be used to move channels from one application
538 to another without needing to exit back into the dialplan. An application must be
539 specified, but the passing a list of arguments to the new application is optional.
540 An example call would look like this:
542 client.channels.move(channelId=chan.id, app='ari-example', appArgs='a,b,c')
544 If the channel was inside of a bridge when switching applications, it will
545 remain there. If the application specified cannot be moved to, then the channel
546 will remain in the current application and an event will be triggered named
547 "ApplicationMoveFailed", which will provide the destination application's name
548 and the channel information.
552 * A new configuration parameter "taskprocessor_overload_trigger" has been
553 added to the pjsip.conf "globals" section. The distributor currently stops
554 accepting new requests when any taskprocessor overload is triggered. The
555 new option allows you to completely disable overload detection (NOT
556 RECOMMENDED), keep the current behavior, or trigger only on pjsip
557 taskprocessor overloads.
561 * A new configuration parameter 'ignore_183_without_sdp' has been added
562 to the pjsip.conf "endpoints" section. If enabled, will make chan_pjsip
563 discard 183s that do not contain an SDP body, which can resolve no
564 ringback tone issues as well as making the behavior match chan_sip.
568 * A new module "res_mwi_devstate" has been added that allows subscriptions
569 to voicemail boxes using "presence" events. This allows common BLF keys
570 to act as voicemail waiting indicators.
574 * Added the ability to set the wrapuptime per-member using the AddQueueMember
577 ------------------------------------------------------------------------------
578 --- Functionality changes from Asterisk 16.1.0 to Asterisk 16.2.0 ------------
579 ------------------------------------------------------------------------------
583 * Whenever an ARI application is started, a context will be created for it
584 automatically as long as one does not already exist, following the format
585 'stasis-<app_name>'. Two extensions are also added to this context: a match-all
586 extension, and the 'h' extension. Any phone that registers under this context
587 will place all calls to the corresponding Stasis application.
591 * Added "send_contact_status_on_update_registration" global configuration option
592 to enable sending AMI ContactStatus event when a device refreshes its registration.
596 * Reworked the media indexer so it doesn't cache the index. Testing revealed
597 that the cache added no benefit but that it could consume excessive memory.
598 Two new index related functions were created: ast_sounds_get_index_for_file()
599 and ast_media_index_update_for_file() which restrict index updating to
600 specific sound files. The original ast_sounds_get_index() and
601 ast_media_index_update() calls are still available but since they no longer
602 cache the results internally, developers should re-use an index they may
603 already have instead of calling ast_sounds_get_index() repeatedly. If
604 information for only a single file is needed, ast_sounds_get_index_for_file()
605 should be called instead of ast_sounds_get_index().
609 * Before Asterisk 12, when using the automon or automixmon features defined
610 in features.conf, a channel variable (TOUCH_MIXMONITOR_OUTPUT) was set on
611 both channels, indicating the filename of the recording.
613 When bridging was overhauled in Asterisk 12, the behavior was changed such
614 that the variable was only set on the peer channel and not on the channel
615 that initiated the automon or automixmon.
617 The previous behavior has been restored so both channels receive the
618 channel variable when one of these features is invoked.
622 * You can now specify a special context with the "aliasescontext" parameter
623 in voicemail.conf which will allow you to create aliases for physical
626 ------------------------------------------------------------------------------
627 --- Functionality changes from Asterisk 16.0.0 to Asterisk 16.1.0 ------------
628 ------------------------------------------------------------------------------
632 * pbx_config will now find and process multiple 'globals' sections from
633 extensions.conf. Variables are processed in the order they are found
634 and duplicate variables overwrite the previous value.
638 * New dialplan function PJSIP_PARSE_URI added to parse an URI and return
639 a specified part of the URI.
643 * ast_bt_get_symbols() now returns a vector of strings instead of an
644 array of strings. This must be freed with ast_bt_free_symbols.
648 * New options 'trust_connected_line' and 'send_connected_line' have been
649 added to the endpoint. The option 'trust_connected_line' is to control
650 if connected line updates are accepted from this endpoint.
651 The option 'send_connected_line' is to control if connected line updates
652 can be sent to this endpoint.
653 The default value is 'yes' for both options.
657 * The existing strictrtp option in rtp.conf has a new choice availabe, called
658 'seqno', which behaves the same way as setting strictrtp to 'yes', but will
659 ignore the time interval during learning so that bursts of packets can still
660 trigger learning our source.
662 ------------------------------------------------------------------------------
663 --- Functionality changes from Asterisk 15 to Asterisk 16 --------------------
664 ------------------------------------------------------------------------------
668 * The app_fax module is now deprecated, users should migrate to the
669 replacement module res_fax.
673 * An 'a' option has been added to the Originate dialplan application which
674 will execute the originate in an asynchronous fashion. If set then the
675 application will return immediately without waiting for the originated
680 * MALLOC_DEBUG no longer has an effect on Asterisk's ABI. Asterisk built
681 with MALLOC_DEBUG can now successfully load binary modules built without
682 MALLOC_DEBUG and vice versa. Third-party pre-compiled modules no longer
683 need to have a special build with it enabled.
685 * Asterisk now depends on libjansson >= 2.11. If this version is not
686 available on your distro you can use `./configure --with-jansson-bundled`.
690 * The app_macro module is now deprecated and by default it is no longer
691 built. Users should migrate to app_stack (Gosub). A warning is logged
692 the first time any Macro is used.
696 * The app_setcallerid module has been removed. The CALLERID dialplan function
697 should be used instead.
701 * New function SIP_HEADERS() enumerates all headers in the incoming INVITE.
703 * The variable GET_TRANSFERRER_DATA set in the peer channel causes matching
704 headers be retrieved from the REFER message and made accessible to the
705 dialplan in the hash TRANSFER_DATA.
709 * Timeouts for reading digits from analog phones are now configurable in
710 chan_dahdi.conf: firstdigit_timeout, interdigit_timeout, matchdigit_timeout.
714 * The ContactStatus and Status fields for the manager events ContactStatus
715 and ContactStatusDetail are now set to "NonQualified" when a contact exists
716 but has not been qualified.
718 * The "Newexten" event is now part of the "dialplan" class. The documentation
719 for Asterisk 15 already specified this, but the implementation was actually
720 using the "call" class instead.
724 * The ContactInfo event's contact_status field is now set to "NonQualified"
725 when a contact exists but has not been qualified.
729 * Added the ability to set the wrapuptime in the configuration of member.
730 When set the wrapuptime on the member is used instead of the wrapuptime
731 defined for the queue itself.
733 * Added predial handler support for caller and callee channels with the
734 B and b options respectively. This is similar to the predial support
739 * The res_config_sqlite module is now deprecated, users should migrate to the
740 replacement module res_config_sqlite3.
744 * The res_monitor module is now deprecated, users should migrate to the
745 replacement module app_mixmonitor.
749 * A new AMI action, PJSIPShowAors, has been added which displays information
750 about all configured PJSIP AORs.
752 * A new AMI action, PJSIPShowAuths, has been added which displays information
753 about all configured PJSIP Auths.
755 * A new AMI action, PJSIPShowContacts, has been added which displays information
756 about all configured PJSIP Contacts.
758 res_pjsip_registrar_expire
760 * The res_pjsip_registrar_expire module has been removed. The functionality has
761 been moved into res_pjsip_registrar.
763 func_audiohookinherit
765 * The func_audiohookinherit module has been removed. Due to architectural changes
766 in Asterisk 12, audiohook inheritance is performed automatically and this
767 function now lacks function.
771 * The cdr_syslog module is now deprecated and by default it is no longer
776 * The cdr_sqlite module has been removed. Users should move to using the
777 cdr_sqlite3_custom module instead.
781 * The format_jpeg module has been removed.
785 * DUNDi now supports IPv6
789 * libedit is no longer available as an embedded library and must be provided
791 * The STATIC_BUILD functionality has been removed as it has not been maintained
792 and has not worked in quite some time.
793 * The module loader now enforces inter-module dependencies. This ensures that
794 a module is not started before another it depends on, even if preload is used.
795 If a dependency is not available or fails to startup this will block any
796 dependants from startup.
797 * Parts of the Asterisk core which can load configuration from realtime are now
798 built-in modules. It is no longer necessary to preload realtime drivers as
799 they are always initialized before the built-in modules.
801 ------------------------------------------------------------------------------
802 --- Functionality changes from Asterisk 15.5.0 to Asterisk 15.6.0 ------------
803 ------------------------------------------------------------------------------
807 * A new option 'suppress_q850_reason_headers' has been added to the endpoint
808 object. Some devices can't accept multiple Reason headers and get confused
809 when both 'SIP' and 'Q.850' Reason headers are received. This option allows
810 the 'Q.850' Reason header to be suppressed. The default value is 'no'.
812 res_pjsip_endpoint_identifier_ip
814 * Added regex support to the identify section match_header option. You
815 specify a regex instead of an explicit string by surrounding the header
817 match_header = SIPHeader: /regex/
819 ------------------------------------------------------------------------------
820 --- Functionality changes from Asterisk 15.4.0 to Asterisk 15.5.0 ------------
821 ------------------------------------------------------------------------------
825 * Core bridging and, more specifically, bridge_softmix have been enhanced to
826 relay received frames of type TEXT or TEXT_DATA to all participants in a
827 softmix bridge. res_pjsip_messaging and chan_pjsip have been enhanced to
828 take advantage of this so when res_pjsip_messaging receives an in-dialog
829 MESSAGE message from a user in a conference call, it's relayed to all
830 other participants in the call.
834 * Support Enhanced Messaging. SendText now accepts new channel variables
835 that can be used to override the To and From display names and set the
836 Content-Type of a message. Since you can now set Content-Type, other
837 text/* content types are now valid.
841 * ConfbridgeList now shows talking status. This utilizes the same voice
842 detection as the ConfbridgeTalking event, so bridges must be configured
843 with "talk_detection_events=yes" for this flag to have meaning.
845 * ConfBridge can now send events to participants via in-dialog MESSAGEs.
846 All current Confbridge events are supported, such as ConfbridgeJoin,
847 ConfbridgeLeave, etc. In addition to those events, a new event
848 ConfbridgeWelcome has been added that will send a list of all
849 current participants to a new participant.
853 * Two new options have been added to the system and endpoint objects to
854 control whether, on outbound calls, Asterisk will accept updated SDP answers
855 during the initial INVITE transaction when 100rel is not in effect.
856 This usually happens when the INVITE is forked to multiple UASs and more
857 than one sends an SDP answer or when a single UAS needs to change a media
858 port to switch from custom ringback to the actual media destination.
860 The 'follow_early_media_forked' option sets whether Asterisk will accept
861 the updated SDP when the To tag on the subsequent response is different than
862 that on the the previous response. This usually occurs in the forked INVITE
863 scenario. The default value is "yes" which is the current behavior.
865 The 'accept_multiple_sdp_answers' flag sets whether Asterisk will accept the
866 updated SDP when the To tag on the subsequent response is the same as that
867 on the previous response. This can occur when a UAS needs to switch media
868 ports from custom ringback to the final media path. The default value is
869 "no" which is the current behavior.
871 These options have to be enabled system-wide in the system config section
872 of pjsip.conf as well as on individual endpoints that require the
875 ------------------------------------------------------------------------------
876 --- Functionality changes from Asterisk 15.3.0 to Asterisk 15.4.0 ------------
877 ------------------------------------------------------------------------------
881 * A new configuration option "genericplc_on_equal_codecs" was added to the
882 "plc" section of codecs.conf to allow generic packet loss concealment even
883 if no transcoding was originally needed. Transcoding via SLIN is forced
888 * Added the "cache_pools" option to pjproject.conf. Disabling the option
889 helps track down pool content mismanagement when using valgrind or
890 MALLOC_DEBUG. The cache gets in the way of determining if the pool contents
891 are used after free and who freed it.
895 * Extend the PJSIPNotify AMI command to send an in-dialog notify on a
898 ------------------------------------------------------------------------------
899 --- Functionality changes from Asterisk 15.2.0 to Asterisk 15.3.0 ------------
900 ------------------------------------------------------------------------------
904 * During dialplan reload log messages are produced for each context,
905 extension and include. These messages are no longer printed by the
906 verbose loggers, they are now only logged as debug messages.
910 * Added the Muted header to the ConfbridgeJoin AMI event to indicate the
911 participant's starting mute status.
913 * Made the AMI ConfbridgeList action's ConfbridgeList events output all
914 the standard channel snapshot headers instead of a few hand-coded channel
915 snapshot headers. The benefit is that the CallerIDName gets disruptive
916 characters like CR, LF, Tab, and a few others escaped. However, an empty
917 CallerIDName is now output as "<unknown>" instead of "<no name>".
921 * Added a new prompt, connecting-prompt, which will be played
922 (if configured) to the "winner" callee before connecting the call.
926 * Users who are matching endpoints by SIP header need to reevaluate their
927 global "endpoint_identifier_order" option in light of the "ip" endpoint
928 identifier method split into the "ip" and "header" endpoint identifier
931 * The pjsip_transport_event feature introduced in 15.1.0 has been refactored.
932 Any external modules that may have used that feature (highly unlikey) will
933 need to be changed as the API has been altered slightly.
935 res_pjsip_endpoint_identifier_ip
937 * The endpoint identifier "ip" method previously recognized endpoints either
938 by IP address or a matching SIP header. The "ip" endpoint identifier method
939 is now split into the "ip" and "header" endpoint identifier methods. The
940 "ip" endpoint identifier method only matches by IP address and the "header"
941 endpoint identifier method only matches by SIP header. The split allows the
942 user to control the relative priority of the IP address and the SIP header
943 identification methods in the global "endpoint_identifier_order" option.
944 e.g., If you have two type=identify sections where one matches by IP address
945 for endpoint alice and the other matches by SIP header for endpoint bob then
946 you can now predict which endpoint is matched when a request comes in that
951 * In an earlier release, inbound registrations on a reliable transport
952 were pruned on Asterisk restart since the TCP connection would have
953 been torn down and become unusable when Asterisk stopped. This same
954 process is now also applied to inbound subscriptions. Since this
955 required the addition of a new column to the ps_subscription_persistence
956 realtime table, users who store their subscriptions in a database will
957 need to run the "alembic upgrade head" process to add the column to
960 res_pjsip_transport_management
962 * Since res_pjsip_transport_management provides several attack
963 mitigation features, its functionality moved to res_pjsip and
964 this module has been removed. This way the features will always
965 be available if res_pjsip is loaded.
967 ------------------------------------------------------------------------------
968 --- Functionality changes from Asterisk 15.1.0 to Asterisk 15.2.0 ------------
969 ------------------------------------------------------------------------------
973 * Added the "cache_media_frames" option to asterisk.conf. Disabling the option
974 helps track down media frame mismanagement when using valgrind or
975 MALLOC_DEBUG. The cache gets in the way of determining if the frame is
976 used after free and who freed it. NOTE: This option has no effect when
977 Asterisk is compiled with the LOW_MEMORY compile time option enabled because
978 the cache code does not exist.
982 * Calls to invalid extensions are now reported as an ACL failure security event
983 "no_extension_match".
987 * The X.509 certificate used for DTLS negotation can now be automatically
988 generated. This is supported by res_pjsip by specifying
989 "dtls_auto_generate_cert = yes" on a PJSIP endpoint. For chan_sip, you
990 would set "dtlsautogeneratecert = yes" either in the [general] section of
991 sip.conf or on a specific peer.
995 * The "identify_by" on endpoints can now be set to "ip" to restrict an endpoint
996 being matched based only on IP address. To ensure no behavior change the
997 default has been changed to "username,ip".
999 ------------------------------------------------------------------------------
1000 --- Functionality changes from Asterisk 15.0.0 to Asterisk 15.1.0 ------------
1001 ------------------------------------------------------------------------------
1005 * The "remove_existing" option now allows a registration to succeed by
1006 displacing any existing contacts that now exceed the "max_contacts" count.
1007 Any removed contacts are the next to expire. The behaviour change is
1008 beneficial when "rewrite_contact" is enabled and "max_contacts" is greater
1009 than one. The removed contact is likely the old contact created by
1010 "rewrite_contact" that the device is refreshing.
1014 * Added a new CancelAtxfer action that cancels an attended transfer.
1016 ------------------------------------------------------------------------------
1017 --- Functionality changes from Asterisk 14 to Asterisk 15 --------------------
1018 ------------------------------------------------------------------------------
1022 * PAUSEALL/UNPAUSEALL now sets the pause reason in the queue_log if it has
1025 * A new option, "announce-position-only-up," has been added that, when set to
1026 yes, causes position announcements to only be played when the caller's
1027 queue position has improved since the last time that we annouced their
1028 position. This default is no.
1032 * '--with-pjproject-bundled' is now the default when running ./configure
1033 It can be disabled with '--without-pjproject-bundled'.
1035 * A '--with-download-cache' option is now available which is equivalent to
1036 setting '--with-sounds-cache' and '--with-externals-cache' to the same
1037 value. The download cache can also be set via the AST_DOWNLOAD_CACHE
1038 environment variable.
1040 ------------------------------------------------------------------------------
1041 --- Functionality changes from Asterisk 14.6.0 to Asterisk 14.7.0 ------------
1042 ------------------------------------------------------------------------------
1046 * The "external_media_address" on transports is now resolved using dnsmgr and
1047 when dnsmgr refreshes are enabled will be automatically updated with the new
1048 IP address of a given hostname.
1050 * A new endpoint parameter "incoming_mwi_mailbox" allows Asterisk to receive
1051 unsolicited MWI NOTIFY requests and make them available to other modules via
1052 the stasis message bus.
1056 * By default, when res_musiconhold reloads or unloads, it sends a HUP signal
1057 to custom applications (and all descendants), waits 100ms, then sends a
1058 TERM signal, waits 100ms, then finally sends a KILL signal. An application
1059 which is interacting with an external device and/or spawns children of its
1060 own may not be able to exit cleanly in the default times, expecially if sent
1061 a KILL signal, or if it's children are getting signals directly from
1062 res_musiconhoild. To allow extra time, the 'kill_escalation_delay'
1063 class option can be used to set the number of milliseconds res_musiconhold
1064 waits before escalating kill signals, with the default being the current
1065 100ms. To control to whom the signals are sent, the "kill_method"
1066 class option can be set to "process_group" (the default, existing behavior),
1067 which sends signals to the application and its descendants directly, or
1068 "process" which sends signals only to the application itself.
1070 * New dialplan function PJSIP_DTMF_MODE added to get or change the DTMF mode
1071 of a channel on a per-call basis.
1075 * OAuth 2.0 authentication is now supported when contacting Google. Follow the
1076 instructions in xmpp.conf.sample to retrieve and configure the necessary
1079 ------------------------------------------------------------------------------
1080 --- Functionality changes from Asterisk 14.5.0 to Asterisk 14.6.0 ------------
1081 ------------------------------------------------------------------------------
1085 * A new global option "imap_poll_logout" was added to specify whether need to
1086 disconnect from the IMAP server after polling of mailboxes.
1091 * A new endpoint option "refer_blind_progress" was added to turn off notifying
1092 the progress details on Blind Transfer. If this option is not set then
1093 the chan_pjsip will send NOTIFY "200 OK" immediately after "202 Accepted".
1094 On default is enabled.
1095 Some SIP phones like Mitel/Aastra or Snom keep the line busy until
1098 * A new endpoint option "notify_early_inuse_ringing" was added to control
1099 whether to notify dialog-info state 'early' or 'confirmed' on Ringing
1102 * The endpoint option 'dtmf_mode' has a new option 'auto_dtmf' added. This
1103 mode works similar to 'auto' except uses DTMF INFO as fallback instead of
1108 * The EAGI() application will now look for a dialplan variable named
1109 EAGI_AUDIO_FORMAT and use that format with the 'enhanced' audio pipe that
1110 EAGI provides. If not specified, it will continue to use the default signed
1115 * When dialing an endpoint directly or using the PJSIP_DIAL_CONTACTS dialplan
1116 function any contact which is considered unreachable due to qualify being
1117 enabled will no longer be called.
1119 * The asymmetric_rtp_codec option now also controls whether chan_pjsip will
1120 send media as-is without transcoding if the codec has been negotiated in the
1121 SDP. If set to "no" then Asterisk will only ever send the preferred codec
1122 from the SDP, unless the remote side sends a different codec and we will
1127 * Added a new PJPROJECT_CONFIGURE_OPTS environment variable which can be used
1128 to pass arbitrary options to the bundled pjproject configure.
1130 * Automatically set the bundled pjproject configure --host and --build
1131 options to match those supplied for the asterisk configure.
1133 ------------------------------------------------------------------------------
1134 --- Functionality changes from Asterisk 14.4.0 to Asterisk 14.5.0 ------------
1135 ------------------------------------------------------------------------------
1139 * Added the stun_blacklist option to rtp.conf. Some multihomed servers have
1140 IP interfaces that cannot reach the STUN server specified by stunaddr.
1141 Blacklist those interface subnets from trying to send a STUN packet to find
1142 the external IP address. Attempting to send the STUN packet needlessly
1143 delays processing incoming and outgoing SIP INVITEs because we will wait
1144 for a response that can never come until we give up on the response.
1145 Multiple subnets may be listed.
1149 * Added logger_queue_limit to the configuration options.
1150 All log messages go to a queue serviced by a single thread
1151 which does all the IO. This setting controls how big that
1152 queue can get (and therefore how much memory is allocated)
1153 before new messages are discarded.
1154 The default is 1000.
1156 res_pjsip_config_wizard
1158 * Two new parameters have been added to the pjsip config wizard.
1159 Setting 'sends_line_with_registrations' to true will cause the wizard
1160 to skip the creation of an identify object to match incoming requests
1161 to the endpoint and instead add the line and endpoint parameters to
1162 the outbound registration object.
1163 Setting 'outbound_proxy' is a shortcut for adding individual
1164 endpoint/outbound_proxy, aor/outbound_proxy and registration/outbound_proxy
1169 * If the 'call-id' value is specified for the uuid_type option and a
1170 chan_sip channel is used the resulting HEP traffic will now contain the
1171 SIP Call-ID instead of the Asterisk channel name.
1173 ------------------------------------------------------------------------------
1174 --- Functionality changes from Asterisk 14.3.0 to Asterisk 14.4.0 ------------
1175 ------------------------------------------------------------------------------
1179 * LOW_MEMORY no longer has an effect on Asterisk ABI. Symbols that were
1180 previously suppressed by LOW_MEMORY are now replaced by stub functions.
1181 Asterisk built with LOW_MEMORY can now successfully load binary modules
1182 built without LOW_MEMORY and vice versa.
1184 * RADIUS backends for CEL and CDR can now also be built using the radcli
1185 client library, in addition to the existing support for building them
1186 using either freeradius or radiusclient-ng.
1190 * ASTERISK_REGISTER_FILE was no longer useful and has been removed. Sources
1191 which use mtx_prof must now manually declare and initialize the variable.
1195 * If an offer is received with optional SRTP (a media stream with RTP/AVP but
1196 which contains a crypto line) chan_sip will now accept it and enable SRTP.
1197 If you would like to do optional SRTP on outbound you will need to create
1198 a dialplan that dials with it enabled initially and if it fails fall back to
1203 * Added endpoint configuration parameter "preferred_codec_only".
1204 This allow asterisk response to a SIP invite with the single most
1205 preferred codec rather than advertising all joint codec capabilities.
1206 This limits the other side's codec choice to exactly what we prefer.
1210 * To fix a memory leak the syslog channel is now empty if it has not been set
1211 and used by a syslog channel in the logger.
1215 * To fix a memory leak the syslog channel is now empty if it has not been set
1216 and used by a syslog channel in the logger.
1220 * New setting "rtp_pt_dynamic = 35" in asterisk.conf:
1221 Normally the Dynamic RTP Payload Type numbers are 96-127, which allow just 32
1222 formats. To avoid the message "No Dynamic RTP mapping available", the range
1223 was changed to 35-63,96-127. This is allowed by RFC 3551 section 3. However,
1224 when you use more than 32 formats and calls are not accepted by a remote
1225 implementation, please report this and go back to rtp_pt_dynamic = 96.
1227 * A new setting, "rtp_use_dynamic", has been added in asterisk.conf". When set
1228 to "yes" RTP dynamic payload types are assigned dynamically per RTP instance.
1229 When set to "no" RTP dynamic payload types are globally initialized to pre-
1230 designated numbers and function similar to static payload types.
1234 * Added support to gosub predial routines on both original channel and on the
1235 created channel using options parameter (like app_dial) B() and b(). This
1236 allows for adding variables to newly created channel or, e.g. setting callerid.
1240 * 'dialplan show' output will now show [config_file:line_number] instead of
1241 [registrar] when that information is available. Currently only extensions
1242 registered by pbx_config when loading/reloading will use this format.
1246 * Add 'QueueUpdate' application which can be used to track outbound calls
1251 * Asterisk will now set the AST_OUTGOING_ATTEMPT channel variable so that
1252 attempt-specific behavior is possible. This is a 1-based number that
1253 simply increases by 1 for each attempt.
1255 ------------------------------------------------------------------------------
1256 --- Functionality changes from Asterisk 14.3.0 to Asterisk 14.4.0 ------------
1257 ------------------------------------------------------------------------------
1261 * The 'PJSIPShowEndpoint' command's respone event of 'IdentifyDetail' now
1262 contains a new optional parameter, 'MatchHeader', mapping to the new
1263 configuration option 'match_header' for the corresponding 'identify' object.
1264 It should be noted that since 'match_header' takes in a key: value pair, the
1265 event parameter will contain a ':' as well.
1269 * Added new 'u' option to Record() application which prevents Asterisk from
1270 truncating silence from the end of recorded files.
1272 res_pjsip_outbound_registration
1274 * Outbound registrations are now refreshed when res_stun_monitor detects
1275 a network change event has happened.
1276 The 'pjsip send (un)register' CLI commands were updated to accept '*all'
1277 as an argument to operate on all registrations.
1278 The 'PJSIP(Un)Register' AMI commands were updated to also accept '*all'.
1282 * The 'Comedian Mail' prompts can now be overriden using the 'vm-login' and
1283 'vm-newuser' configuration options in voicemail.conf.
1285 * Added 'fromstring' field to the voicemail boxes. If set, it will override
1286 the global 'fromstring' field on a per-mailbox basis.
1290 * Added CHANNEL(callid) to retrieve the call log tag associated with the
1291 channel. e.g., [C-00000000] Dialplan now has access to the call log
1292 search key associated with the channel so it can be saved in case there
1293 is a problem with the call.
1297 * A new transport parameter 'symmetric_transport' has been added.
1298 When a request from a dynamic contact comes in on a transport with this
1299 option set to 'yes', the transport name will be saved and used for
1300 subsequent outgoing requests like OPTIONS, NOTIFY and INVITE. It's
1301 saved as a contact uri parameter named 'x-ast-txp' and will display with
1302 the contact uri in CLI, AMI, and ARI output. On the outgoing request,
1303 if a transport wasn't explicitly set on the endpoint AND the request URI
1304 is not a hostname, the saved transport will be used and the 'x-ast-txp'
1305 parameter stripped from the outgoing packet. To facilitate recreation of
1306 subscriptions on asterisk restart, a new column 'contact_uri' needed to be
1307 added to the ps_subcsription_persistence table. Since new columns were
1308 added to both transport and subscription_persistence, an alembic upgrade
1309 should be run to bring the database tables up to date.
1311 * A new option, allow_overlap, has been added to endpoints which allows
1312 overlap dialing functionality to be enabled or disabled. The option defaults
1315 res_pjsip_transport_websocket
1317 * Removed non-secure websocket support. Firefox and Chrome have not allowed
1318 non-secure websockets for quite some time so this shouldn't be an issue
1319 for people. Attempting to use a non-secure websocket may or may not work
1320 when Asterisk attempts to send SIP requests to do something like initiate
1323 res_pjsip_endpoint_identifier_ip
1325 * A new option has been added to the 'identify' configuration object,
1326 'match_header'. The 'match_header' attribute should contain a SIP
1327 header: value pair that, When set, will cause inbound requests that contain
1328 the matching SIP header/value pair to be associated with the corresponding
1329 endpoint. This option is cumulative with the 'match' option, so that if
1330 either option matches the request, the request is associated with the
1333 In a future release, this module will be renamed to something more
1334 appropriate, as it now matches inbound requests on more than just IP
1339 * The RTP layer of Asterisk now has support for RFC 5761: "Multiplexing RTP
1340 Data and Control Packets on a Single Port." So far, the only channel driver
1341 that supports this feature is chan_pjsip. You can set "rtcp_mux = yes" on
1342 a PJSIP endpoint in pjsip.conf to enable the feature.
1344 ------------------------------------------------------------------------------
1345 --- Functionality changes from Asterisk 14.2.0 to Asterisk 14.3.0 ------------
1346 ------------------------------------------------------------------------------
1350 * Added new CLI command "pjproject set log level". The new command allows
1351 the maximum PJPROJECT log levels to be adjusted dynamically and
1352 independently from the set debug logging level like many other similar
1353 module debug logging commands.
1355 * Added new companion CLI command "pjproject show log level" to allow the
1356 user to see the current maximum pjproject logging level.
1358 * Added new pjproject.conf startup section "log_level' option to set the
1359 initial maximum PJPROJECT logging level.
1361 res_pjsip_outbound_registration
1363 * Statsd no longer logs redundant status PJSIP.registrations.state changes
1364 for internal state transitions that don't change the reported public status
1369 * The PJSIPShowRegistrationInboundContactStatuses AMI command has been added
1370 to return ContactStatusDetail events as opposed to
1371 PJSIPShowRegistrationsInbound which just a dumps every defined AOR.
1375 * Six existing contact fields have been added to the end of the
1376 ContactStatusDetail AMI event:
1377 ID, AuthenticateQualify, OutboundProxy, Path, QualifyFrequency and
1378 QualifyTimeout. Existing fields have not been disturbed.
1380 res_pjsip_endpoint_identifier_ip
1382 * SRV lookups can now be done on provided hostnames to determine additional
1383 source IP addresses for requests. This is configurable using the
1384 "srv_lookups" option on the identify and defaults to "yes".
1388 * The 'ari set debug' command has been enhanced to accept 'all' as an
1389 application name. This allows dumping of all apps even if an app
1390 hasn't registered yet.
1392 * 'ari set debug' now displays requests and responses as well as events.
1394 ------------------------------------------------------------------------------
1395 --- Functionality changes from Asterisk 14.1.0 to Asterisk 14.2.0 ------------
1396 ------------------------------------------------------------------------------
1400 * Events that reference a bridge may now contain two new optional fields:
1401 - 'BridgeVideoSourceMode': the video source mode for the bridge.
1402 Can be one of 'none', 'talker', or 'single'.
1403 - 'BridgeVideoSource': the unique ID of the channel that is the video
1404 source in this bridge, if one exists.
1406 * A new event, BridgeVideoSourceUpdate, has been added with a class
1407 authorization of CALL. The event is raised when the video source changes
1408 in a multi-party mixing bridge.
1412 * The bridges resource now exposes two new operations:
1413 - POST /bridges/{bridgeId}/videoSource/{channelId}: Set a video source in a
1414 multi-party mixing bridge
1415 - DELETE /bridges/{bridgeId}/videoSource: Remove the set video source,
1416 reverting to talk detection for the video source
1418 * The bridge model in any returned response or event now contains the following
1420 - video_mode: the video source mode for the bridge. Can be one of 'none',
1421 'talker', or 'single'.
1422 - video_source_id: the unique ID of the channel that is the video source
1423 in this bridge, if one exists.
1425 * A new event, BridgeVideoSourceChanged, has been added for bridges.
1426 Applications subscribed to a bridge will receive this event when the source
1427 of video changes in a mixing bridge.
1429 * The ARI major version has been bumped. There are not any known breaking changes
1430 in ARI. The major version has been bumped because otherwise we can end up with
1431 overlapping version numbers between different Asterisk versions. Now each major
1432 version of Asterisk will bring with it a change in the major version of ARI.
1433 The ARI version in Asterisk 14 is now 2.0.0.
1437 * Automatic dual stack support is now implemented. Depending on DNS resolution
1438 and the transport used for sending a message the SIP signaling and SDP will
1439 be updated with the correct IP address and protocol version. This means that
1440 the rtp_ipv6 and t38_udptl_ipv6 options no longer have any effect. The
1441 res_pjsip_multihomed module has also been moved into core res_pjsip to ensure
1442 that messages are updated with the correct address information in all cases.
1446 * The default behavior for RTP codecs has been changed. The sending codec will
1447 now match the receiving codec. This can be turned off and behavior reverted
1448 to asymmetric using the "asymmetric_rtp_codec" endpoint option. If this
1449 option is set then the sending and received codec are allowed to differ.
1453 * Three new CLI commands have been added for ARI:
1455 Displays a listing of all registered ARI applications.
1456 - ari show app <name>:
1457 Display detailed information about a registered ARI application.
1458 - ari set debug <name> <on|off>:
1459 Enable/disable debugging of an ARI application. When debugged, verbose
1460 information will be sent to the Asterisk CLI.
1465 * A new dialplan variable, ABANDONED, is set when the call is not answered
1470 * The configuration file ari.conf now supports a channelvars option, which
1471 specifies a list of channel variables to include in each channel-oriented
1474 ------------------------------------------------------------------------------
1475 --- Functionality changes from Asterisk 14.0.0 to Asterisk 14.1.0 ------------
1476 ------------------------------------------------------------------------------
1480 * The res_digium_phone, codec_g729a, codec_silk, codec_siren7 and
1481 codec_siren14 binary modules hosted at downloads.digium.com can now be
1482 automatically downloaded and installed during the Asterisk install
1483 process. If selected in menuselect, when 'make install' is run, the
1484 script will check the downloads site for a new version and download
1485 and install it if needed. The '--with-externals-cache' option to
1486 ./configure can be used to specify a location to cache the latest
1487 tarballs so they don't have to be re-downloaded for every install.
1491 * Added "tps_queue_high" and "tps_queue_low" options.
1492 The options can modify the taskprocessor alert levels for this module.
1493 Additional information can be found in the sample configuration file at
1494 config/samples/voicemail.conf.sample.
1498 * Added "mwi_tps_queue_high" and "mwi_tps_queue_low" global configuration
1499 options to tune taskprocessor alert levels.
1501 * Added "mwi_disable_initial_unsolicited" global configuration option
1502 to disable sending unsolicited MWI to all endpoints on startup.
1503 Additional information can be found in the sample configuration file at
1504 config/samples/pjsip.conf.sample.
1508 * A new dialplan function, PJSIP_SEND_SESSION_REFRESH, has been added. When
1509 invoked, a re-INVITE or UPDATE request will be sent immediately to the
1510 endpoint underlying the channel. When used in combination with the existing
1511 dialplan function PJSIP_MEDIA_OFFER, this allows the formats on a PJSIP
1512 channel to be re-negotiated and updated after session set up.
1516 * A new endpoint configuration parameter 'contact_user' has been added which
1517 when set will override the default user set on Contact headers in outgoing
1520 * If you are using a sorcery realtime backend to store global res_pjsip
1521 options (ps_globals table) then you now have to do a res_pjsip reload for
1522 changes to these options to take effect. If you are using pjsip.conf to
1523 configure these options then you already had to do a reload after making
1526 * Added "ignore_uri_user_options" global configuration option for
1527 compatibility with an ITSP that sends URI user field options. When enabled
1528 the user field is truncated at the first semicolon.
1530 URI: "sip:1235557890;phone-context=national@x.x.x.x;user=phone"
1531 The user field is "1235557890;phone-context=national"
1532 Which is truncated to this: "1235557890"
1534 Note: The caller-id and redirecting number strings obtained from incoming
1535 SIP URI user fields are now always truncated at the first semicolon.
1539 * An option, ice_blacklist, has been added which allows certain subnets to be
1540 excluded from local ICE candidates.
1544 * Some sounds played into the bridge are played asynchronously. This, for
1545 instance, allows a channel to immediately exit the ConfBridge without having
1546 to wait for a leave announcement to play.
1550 * Added the "Q" option which sets the Q.850/Q.931 cause on unanswered channels
1551 when another channel answers the call. The default of ANSWERED_ELSEWHERE
1556 * ARI events will all now include a new field in the root of the JSON message,
1557 'asterisk_id'. This will be the unique ID for the Asterisk system
1558 transmitting the event. The value can be overridden using the 'entityid'
1559 setting in asterisk.conf.
1561 ------------------------------------------------------------------------------
1562 --- Functionality changes from Asterisk 13 to Asterisk 14 --------------------
1563 ------------------------------------------------------------------------------
1567 * A new event, "DialState" has been added. This is similar to "DialBegin" and
1568 "DialEnd" in that it tracks the state of a dialed call. The difference is that
1569 this indicates some intermediate state change in the dial attempt, such as
1570 "RINGING", "PROGRESS", or "PROCEEDING".
1574 * A new ARI method has been added to the channels resource. "create" allows for
1575 you to create a new channel and place that channel into a Stasis application.
1576 This is similar to origination except that the specified channel is not
1577 dialed. This allows for an application writer to create a channel, perform
1578 manipulations on it, and then delay dialing the channel until later.
1580 * To complement the "create" method, a "dial" method has been added to the
1581 channels resource in order to place a call to a created channel.
1583 * All operations that initiate playback of media on a resource now support
1584 a list of media URIs. The list of URIs are played in the order they are
1585 presented to the resource. A new event, "PlaybackContinuing", is raised when
1586 a media URI finishes but before the next media URI starts. When a list is
1587 played, the "Playback" model will contain the optional attribute
1588 "next_media_uri", which specifies the next media URI in the list to be played
1589 back to the resource. The "PlaybackFinished" event is raised when all media
1592 * Stored recordings now allow for the media associated with a stored recording
1593 to be retrieved. The new route, GET /recordings/stored/{name}/file, will
1594 transmit the raw media file to the requester as binary.
1597 * "Dial" events have been modified to not only be sent when dialing begins and ends.
1598 They now are also sent for intermediate states, such as "RINGING", "PROGRESS", and
1606 * A new application in Asterisk, this will join the calling channel
1607 to an existing bridge containing the named channel prefix.
1611 * Added the 'l' option, which forces ChanSpy's audiohook to use a long queue
1612 to store the audio frames. This option is useful if audio loss is
1613 experienced when using ChanSpy, but may introduce some delay in the audio
1614 feed on the listening channel.
1618 * Added format attribute negotiation for the iLBC audio codec. Format attribute
1619 negotiation is provided by the res_format_attr_ilbc module. iLBC 20 is the
1620 default now. Falls back to iLBC 30, when the remote party requests this.
1624 * Added the ability to pass options to MixMonitor when recording is used with
1625 ConfBridge. This includes the addition of the following configuration
1626 parameters for the 'bridge' object:
1627 - record_file_timestamp: whether or not to append the start time to the
1629 - record_options: the options to pass to the MixMonitor application
1630 - record_command: a command to execute when recording is finished
1631 Note that these options may also be with the CONFBRIDGE function.
1635 * Remote files can now be retrieved and played back. See the Playback
1636 dialplan application for more details.
1640 * It is now possible to disable the prompt from a callee by setting
1641 'enable_callee_prompt = no' in followme.conf.
1645 * Remote files can now be retrieved and played back via the Playback and other
1646 media playback dialplan applications. This is done by directly providing
1647 the URL to play to the dialplan application:
1648 same => n,Playback(http://1.1.1.1/howler-monkeys-fl.wav)
1649 Note that unlike 'normal' media files, the entire URI to the file must be
1650 provided, including the file extension. Currently, on HTTP and HTTPS URI
1651 schemes are supported.
1655 * Added field ReasonPause on QueueMemberStatus if set when paused, the reason
1656 the queue member was paused.
1658 * Added field LastPause on QueueMemberStatus for time when started the last
1659 pause for a queue member.
1661 * Show the time when started the last pause for queue member on CLI for command
1666 * Added the 'n' option, which prevents the SMS from being written to the log
1667 file. This is needed for those countries with privacy laws that require
1668 providers to not log SMS content.
1676 * The CALLERID(ani2) value for incoming calls is now populated in featdmf
1677 signaling mode. The information was previously discarded.
1679 * Added the force_restart_unavailable_chans compatibility option. When
1680 enabled it causes Asterisk to restart the ISDN B channel if an outgoing
1681 call receives cause 44 (Requested channel not available).
1685 * The iax.conf forcejitterbuffer option has been removed. It is now always
1686 forced if you set iax.conf jitterbuffer=yes. If you put a jitter buffer
1687 on a channel it will be on the channel.
1689 * A new configuration parameters, 'calltokenexpiration', has been added that
1690 controls the duration before a call token expires. Default duration is 10
1691 seconds. Setting this to a higher value may help in lagged networks or those
1692 experiencing high packet loss.
1694 * Plaintext auth mode is deprecated and removed from possible default modes.
1696 chan_rtp (was chan_multicast_rtp)
1698 * Added unicast RTP support and renamed chan_multicast_rtp to chan_rtp.
1700 * The format for dialing a unicast RTP channel is:
1701 UnicastRTP/<destination-addr>[/[<options>]]
1702 Where <destination-addr> is something like '127.0.0.1:5060'.
1703 Where <options> are in standard Asterisk flag options format:
1704 c(<codec>) - Specify which codec/format to use such as 'ulaw'.
1705 e(<engine>) - Specify which RTP engine to use such as 'asterisk'.
1707 * New options were added for a multicast RTP channel. The format for
1708 dialing a multicast RTP channel is:
1709 MulticastRTP/<type>/<destination-addr>[/[<control-addr>][/[<options>]]]
1710 Where <type> can be either 'basic' or 'linksys'.
1711 Where <destination-addr> is something like '224.0.0.3:5060'.
1712 Where <control-addr> is something like '127.0.0.1:5060'.
1713 Where <options> are in standard Asterisk flag options format:
1714 c(<codec>) - Specify which codec/format to use such as 'ulaw'.
1715 i(<address>) - Specify the interface address from which multicast RTP
1717 l(<enable>) - Set whether packets are looped back to the sender. The
1718 enable value can be 0 to set looping to off and non-zero to set
1720 t(<ttl>) - Set the time-to-live (TTL) value for multicast packets.
1724 * New 'rtpbindaddr' global setting. This allows a user to define which
1725 ipaddress to bind the rtpengine to. For example, chan_sip might bind
1726 to eth0 (10.0.0.2) but rtpengine to eth1 (192.168.1.10).
1728 * DTLS related configuration options can now be set at a general level.
1729 Enabling DTLS support, though, requires enabling it at the user
1732 * Added the possibility to set the From: header through the the SIP dial
1733 string (populating the fromuser/fromdomain fields), complementing the
1734 [!dnid] option for the To: header that has existed since 1.6.0 (1d6b192).
1735 NOTE: This is again separated by an exclamation mark, so the To: header may
1736 not contain one of those.
1738 * Session-Timers (RFC 4028) work for TCP (and TLS) transports as well now.
1739 Previously Asterisk dropped calls only with UDP transports. However with
1740 longer international calls via TCP, the SIP channel might break, because
1741 all hops on the Internet route must stay online (have not a single power
1742 outage, for example). Therefore with Session-Timers enabled (which are
1743 enabled at default), you might see additional dropped calls. Consequently
1744 please, consider to go for session-timers=refuse in your sip.conf.
1748 * New 'user_eq_phone' endpoint setting. This adds a 'user=phone' parameter
1749 to the request URI and From URI if the user is determined to be a phone
1752 * New 'moh_passthrough' endpoint setting. This will pass hold and unhold
1753 requests through using SIP re-invites with sendonly and sendrecv accordingly.
1755 * Added the pjsip.conf system type disable_tcp_switch option. The option
1756 allows the user to disable switching from UDP to TCP transports described
1757 by RFC 3261 section 18.1.1.
1759 * New 'line' and 'endpoint' options added on outbound registrations. This
1760 allows some identifying information to be added to the Contact of the
1761 outbound registration. If this information is present on messages received
1762 from the remote server the message will automatically be associated with the
1763 configured endpoint on the outbound registration.
1768 * The core of Asterisk uses a message bus called "Stasis" to distribute
1769 information to internal components. For performance reasons, the message
1770 distribution was modified to make use of a thread pool instead of a
1771 dedicated thread per consumer in certain cases. The initial settings for
1772 the thread pool can now be configured in 'stasis.conf'.
1774 * A new core DNS API has been implemented which provides a common interface
1775 for DNS functionality. Modules that use this functionality will require that
1776 a DNS resolver module is loaded and available.
1778 * Modified processing of command-line options to first parse only what
1779 is necessary to read asterisk.conf. Once asterisk.conf is fully loaded,
1780 the remaining options are processed. The -X option now applies to
1781 asterisk.conf only. To enable #exec for other config files you must
1782 set execincludes=yes in asterisk.conf. Any other option set on the
1783 command-line will now override the equivalent setting from asterisk.conf.
1785 * The TLS core in Asterisk now supports X.509 certificate subject alternative
1786 names. This way one X.509 certificate can be used for hosts that can be
1787 reached under multiple DNS names or for multiple hosts.
1789 * The Asterisk logging system now supports JSON structured logging. Log
1790 channels specified in logger.conf or added dynamically via CLI commands now
1791 support an optional specifier prior to their levels that determines their
1792 formatting. To set a log channel to format its entries as JSON, a formatter
1793 of '[json]' can be set, e.g.,
1794 full => [json]debug,verbose,notice,warning,error
1796 * The core now supports a 'media cache', which stores temporary media files
1797 retrieved from external sources. CLI commands have been added to manipulate
1798 and display the cached files, including:
1799 - 'media cache show <all>' - show all cached media files, or details about
1800 one particular cached media file
1801 - 'media cache refresh <item>' - force a refresh of a particular media file
1803 - 'media cache delete <item>' - remove an item from the cache
1804 - 'media cache create <uri>' - retrieve a URI and store it in the cache
1806 * The ability for device state hints to be automatically created as a result of
1807 device state changes now exists in the PBX. This functionality is referred to
1808 as "autohints" and is configurable in extensions.conf by placing "autohints=yes"
1809 in the context. If enabled a device state hint will be automatically created
1810 with the name of the device.
1812 * If Asterisk is built with systemd support, and run under systemd, it will
1813 notify systemd of its state using sd_notify. Use 'Type=notify' in
1818 * The func_odbc global option "single_db_connection" default value has been
1824 * New module format_ogg_speex added which supports Speex codec inside
1825 Ogg containers (filename extension .spx).
1830 * Added CHANNEL(onhold) item that returns 1 (onhold) and 0 (not-onhold) for
1831 the hold status of a channel.
1835 * The CURL function now supports a write option, which will save the retrieved
1836 file to a location on disk. As an example:
1837 same => n,Set(CURL(https://1.1.1.1/foo.wav)=/tmp/foo.wav)
1838 will save 'foo.wav' to /tmp.
1842 * The transferdialattempts default value has been changed from 1 to 3. The
1843 transferinvalidsound has been changed from "pbx-invalid" to
1844 "privacy-incorrect". These were changed to make DTMF transfers be more
1845 user-friendly by default.
1851 res_http_media_cache
1853 * A backend for the core media cache, this module retrieves media files from
1854 a remote HTTP(S) server and stores them in the core media cache for later
1859 * Added sort=randstart to the sort options. It sorts the files by name and
1860 then chooses the first file to play at random.
1861 * Added preferchannelclass=no option to prefer the application-passed class
1862 over the channel-set musicclass. This allows separate hold-music from
1863 application (e.g. Queue or Dial) specified music.
1865 res_resolver_unbound
1867 * Added a res_resolver_unbound module which uses the libunbound resolver library
1868 to perform DNS resolution. This module requires the libunbound library to be
1869 installed in order to be used.
1873 * A new SIP resolver using the core DNS API has been implemented. This relies on
1874 external SIP resolver support in PJSIP which is only available as of PJSIP
1875 2.4. If this support is unavailable the existing built-in PJSIP SIP resolver
1876 will be used instead. The new SIP resolver provides NAPTR support, improved
1877 SRV support, and AAAA record support.
1879 res_pjsip_info_empty
1880 --------------------
1881 * A new module that can respond to empty Content-Type INFO packets during call.
1882 Some SBCs will terminate a call if their empty INFO packets are not responded
1883 to within a predefined time.
1885 res_pjsip_outbound_registration
1886 -------------------------------
1887 * A new 'fatal_retry_interval' option has been added to outbound registration.
1888 When set (default is zero), and upon receiving a failure response to an
1889 outbound registration, registration is retried at the given interval up to
1892 res_pjsip_outbound_publish
1894 * Added a new multi_user option that when set to 'yes' allows a given configuration
1895 to be used for multiple users.
1903 * Added a new option, 'usegmtime', which causes timestamps in CEL events
1904 to be logged in GMT.
1906 * Added support to set schema where located the table cel. This settings is
1907 configurable for cel_pgsql via the 'schema' in configuration file
1916 * Added the ability to set the character to quote identifiers. This
1917 allows adding the character at the start and end of table and column
1918 names. This setting is configurable for cdr_adaptive_odbc via the
1919 quoted_identifiers in configuration file cdr_adaptive_odbc.conf.
1923 * Added a new configuration option, "newcdrcolumns", which enables use of the
1924 post-1.8 CDR columns 'peeraccount', 'linkedid', and 'sequence'.
1928 * Added a new configuration option, "newcdrcolumns", which enables use of the
1929 post-1.8 CDR columns 'peeraccount', 'linkedid', and 'sequence'.
1931 ------------------------------------------------------------------------------
1932 --- Functionality changes from Asterisk 13.10.0 to Asterisk 13.11.0 ----------
1933 ------------------------------------------------------------------------------
1937 * Added "faxdetect_timeout" option.
1938 The option determines how many seconds into a call before faxdetect
1939 is disabled for the call. Setting the value to zero disables the timeout.
1943 * Added "fax_detect_timeout" to endpoint.
1944 The option determines how many seconds into a call before fax_detect
1945 is disabled for the call. Setting the value to zero disables the timeout.
1947 * Added "subscribe_context" to endpoint.
1948 If specified, incoming SUBSCRIBE requests will be searched for the matching
1949 extension in the indicated context. If no "subscribe_context" is specified,
1950 then the "context" setting is used.
1954 * The DTLS part in Asterisk now supports Perfect Forward Secrecy (PFS).
1955 Enabling PFS is attempted by default, and is dependent on the configuration
1956 of the module using TLS.
1957 - Ephemeral ECDH (ECDHE) is enabled by default. To disable it, do not
1958 specify a ECDHE cipher suite in sip.conf, for example:
1959 dtlscipher=AES128-SHA
1960 - Ephemeral DH (DHE) is disabled by default. To enable it, add DH parameters
1961 into the private key file, e.g., sip.conf dtlsprivatekey. For example:
1962 openssl dhparam -out ./dh.pem 2048
1963 - Because clients expect the server to prefer PFS, and because OpenSSL sorts
1964 its cipher suites by bit strength, see "openssl ciphers -v DEFAULT".
1965 Consider re-ordering your cipher suites in the respective configuration
1967 dtlscipher=ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-RSA-AES128-GCM-SHA256
1968 which forces PFS and requires at least DTLS 1.2.
1970 ------------------------------------------------------------------------------
1971 --- Functionality changes from Asterisk 13.9.0 to Asterisk 13.10.0 -----------
1972 ------------------------------------------------------------------------------
1976 * A channel variable FORWARDERNAME is now set which indicates which channel
1977 was responsible for a forwarding requests received on dial attempt.
1981 * Added new global option "single_db_connection".
1982 Enabling this option func_odbc will use a single database connection per DSN.
1983 This option is enabled by default.
1987 * Added FAXMODE variable to let dialplan know what fax transport was used.
1988 FAXMODE variable is set to either "audio" or "T38".
1992 * Added "via_addr", "via_port", "call_id" to contacts.
1993 As res_pjsip_nat rewrites contact's address, only the last Via header
1994 can contain the source address of registered endpoint.
1995 Also Call-Id header may contain the source address of registered endpoint.
1996 Added new fields ViaAddress,CallID to AMI event ContactStatus
1998 * Endpoint IP Access Controls
1999 Added new configuration Endpoint options:
2000 "acl" - list of IP ACL section names in acl.conf
2001 "deny" - List of IP addresses to deny access from
2002 "permit" - List of IP addresses to permit access from
2003 "contact_acl" - List of Contact ACL section names in acl.conf
2004 "contact_deny" - List of Contact header addresses to deny
2005 "contact_permit" - List of Contact header addresses to permit
2007 * Added "reg_server" to contacts.
2008 If the Asterisk system name is set in asterisk.conf, it will be stored
2009 into the "reg_server" field in the ps_contacts table to facilitate
2010 multi-server setups.
2012 * When starting Asterisk, received traffic will now be ignored until Asterisk
2013 has loaded all modules and is fully booted.
2017 * Added a new option, 'uuid_type', that sets the preferred source of the Homer
2018 correlation UUID. The valid options are:
2019 - call-id: Use the PJSIP SIP Call-ID header value
2020 - channel: Use the Asterisk channel name
2021 The default value is 'call-id'. In the event that a HEP module cannot find a
2022 valid value using the specified 'uuid_type', the module may fallback to a
2023 more readily available source for the correlation UUID.
2027 * A new option has been added, 'max_connections', which sets the maximum number
2028 of concurrent connections to the database. This option defaults to 1 which
2029 returns the behavior to that of Asterisk 13.7 and prior.
2033 * Added a bridge profile option called regcontext that allows you to
2034 dynamically register the conference bridge name as an extension into
2035 the specified context. This allows tracking down conferences on multi-
2036 server installations via alternate means (DUNDI for example). By default
2037 this feature is not used.
2041 * Added the associated format name to 'core show codecs'.
2045 * Added 'formats' to channel create/originate to allow setting the allowed
2046 formats for a channel when no originator channel is available. Especially
2047 useful for Local channel creation where no other format information is
2048 available. 'core show codecs' can now be used to look up suitable format
2051 ------------------------------------------------------------------------------
2052 --- Functionality changes from Asterisk 13.8.0 to Asterisk 13.9.0 ------------
2053 ------------------------------------------------------------------------------
2056 - The dynamic parking lot creation channel variables PARKINGDYNAMIC,
2057 PARKINGDYNCONTEXT, PARKINGDYNEXTEN, and PARKINGDYNPOS are now looked
2058 for in the parker's channel instead of the parked channel. This is only
2059 of significance if the parker uses blind transfer or the DTMF one-step
2060 parking feature. You need to use the double underscore '__' inheritance
2061 for these variables. The indefinite inheritance is also recommended
2062 for the PARKINGEXTEN variable.
2066 * Added new global option (disable_multi_domain) to pjsip.
2067 Disabling Multi Domain can improve realtime performace by reducing
2068 number of database requsts.
2072 * Added 'pjsip show channelstats' CLI command.
2074 res_pjsip_outbound_publish
2076 * Added support for setting the transport used on outbound publish
2077 using the transport configuration option.
2079 ------------------------------------------------------------------------------
2080 --- Functionality changes from Asterisk 13.7.0 to Asterisk 13.8.0 ------------
2081 ------------------------------------------------------------------------------
2085 * Per RFC3325, the 'From' header is now anonymized on outgoing calls when
2086 caller id presentation is prohibited.
2088 res_pjsip_config_wizard
2090 * A new command (pjsip export config_wizard primitives) has been added that
2091 will export all the pjsip objects it created to the console or a file
2092 suitable for reuse in a pjsip.conf file.
2096 * To help insure that Asterisk is compiled and run with the same known
2097 version of pjproject, a new option (--with-pjproject-bundled) has been
2098 added to ./configure. When specified, the version of pjproject specified
2099 in third-party/versions.mak will be downloaded and configured. When you
2100 make Asterisk, the build process will also automatically build pjproject
2101 and Asterisk will be statically linked to it. Once a particular version
2102 of pjproject is configured and built, it won't be configured or built
2103 again unless you run a 'make distclean'.
2105 To facilitate testing, when 'make install' is run, the pjsua and pjsystest
2106 utilities and the pjproject python bindings will be installed in
2107 ASTDATADIR/third-party/pjproject.
2109 The default behavior remains building with the shared pjproject
2110 installation, if any.
2114 * Added CONFBRIDGE_INFO(muted,) for querying the muted conference state.
2116 * Added Muted header to AMI ConfbridgeListRooms action response list events
2117 to indicate the muted conference state.
2119 * Added Muted column to CLI "confbridge list" output to indicate the muted
2120 conference state and made the locked column a yes/no value instead of a
2121 locked/unlocked value.
2125 * The REDIRECTING(reason) value is now treated consistently between
2126 chan_sip and chan_pjsip.
2128 Both channel drivers match incoming reason values with values documented
2129 by REDIRECTING(reason) and values documented by RFC5806 regardless of
2130 whether they are quoted or not. RFC5806 values are mapped to the
2131 equivalent REDIRECTING(reason) documented value and is set in
2132 REDIRECTING(reason). e.g., an incoming RFC5806 'unconditional' value or a
2133 quoted string version ('"unconditional"') is converted to
2134 REDIRECTING(reason)'s 'cfu' value. The user's dialplan only needs to deal
2135 with 'cfu' instead of any of the aliases.
2137 The incoming 480 response reason text supported by chan_sip checks for
2138 known reason values and if not matched then puts quotes around the reason
2139 string and assigns that to REDIRECTING(reason).
2141 Both channel drivers send outgoing known REDIRECTING(reason) values as the
2142 unquoted RFC5806 equivalent. User custom values are either sent as is or
2143 with added quotes if SIP doesn't allow a character within the value as
2144 part of a RFC3261 Section 25.1 token. Note that there are still
2145 limitations on what characters can be put in a custom user value. e.g.,
2146 embedding quotes in the middle of the reason string is just going to cause
2149 * Setting a REDIRECTING(reason) value now recognizes RFC5806 aliases.
2150 e.g., Setting REDIRECTING(reason) to 'unconditional' is converted to the
2155 * This module is the successor of res_pjsip_log_forwarder. As well as
2156 handling the log forwarding (which now displays as 'pjproject:0' instead
2157 of 'pjsip:0'), it also adds a 'pjproject show buildopts' command to the CLI.
2158 This displays the compiled-in options of the pjproject installation
2159 Asterisk is currently running against.
2161 * Another feature of this module is the ability to map pjproject log levels
2162 to Asterisk log levels, or to suppress the pjproject log messages
2163 altogether. Many of the messages emitted by pjproject itself are the result
2164 of errors which Asterisk will ultimately handle so the messages can be
2165 misleading or just noise. A new config file (pjproject.conf) has been added
2166 to configure the mapping and a new CLI command (pjproject show log mappings)
2167 has been added to display the mappings currently in use.
2171 * Transports are now reloadable. In testing, no in-progress calls were
2172 disrupted if the ip address or port weren't changed, but the possibility
2173 still exists. To make sure there are no unintentional drops, a new option
2174 'allow_reload', which defaults to 'no' has been added to transport. If
2175 left at the default, changes to the particular transport will be ignored.
2176 If set to 'yes', changes (if any) will be applied.
2178 * Added new global option (regcontext) to pjsip. When set, Asterisk will
2179 dynamically create and destroy a NoOp priority 1 extension
2180 for a given endpoint who registers or unregisters with us.
2182 * Endpoints and aors can now be identified by the username and realm in an
2183 incoming Authorization header. To use this feature, add "auth_username"
2184 to your endpoint's "identify_by" list. You can combine "auth_username"
2185 and the original "username" to test both the From/To and Authorization
2186 headers. For endpoints, the order is controlled by the global
2187 "endpoint_identifier_order" setting. For matching aors to an endpoint
2188 for inbound registration, the order is controlled by this option.
2190 * In conjunction with the "auth_username" change, 3 new options have been
2191 added to the global configuration object that control how many unidentified
2192 requests over a certain period from the same IP address can be received
2193 before a security altert is generated. A new CLI command
2194 "pjsip show unidentified_requests" will list the current candidates.
2198 * A new module, res_pjsip_history, has been added that provides SIP history
2199 viewing/filtering from the CLI. The module is intended to be used on systems
2200 with busy SIP traffic, where existing forms of viewing SIP messages - such
2201 as the res_pjsip_logger - may be inadequate. The module provides two new
2203 - 'pjsip set history {on|off|clear}' - this enables/disables SIP history
2204 capturing, as well as clears an existing history capture. Note that SIP
2205 packets captured are stored in memory until cleared. As a result, the
2206 history capture should only be used for debugging/viewing purposes, and
2207 should *NOT* be left permanently enabled on a system.
2208 - 'pjsip show history' - displays the captured SIP history. When invoked
2209 with no options, the entire captured history is displayed. Two options
2211 -- 'entry <num>' - display a detailed view of a single SIP message in
2213 -- 'where ...' - filter the history based on some expression. For more
2214 information on filtering, view the current CLI help for the
2215 'pjsip show history' command.
2219 * app_voicemail and res_mwi_external can now be built together. The default
2220 remains to build app_voicemail and not res_mwi_external but if they are
2221 both built, the load order will cause res_mwi_external to load first and
2222 app_voicemail will be skipped. Use 'preload=app_voicemail.so' in
2223 modules.conf to force app_voicemail to be the voicemail provider.
2227 * A new option (bind_rtp_to_media_address) has been added to endpoint which
2228 will cause res_pjsip_sdp_rtp to actually bind the RTP instance to the
2229 media_address as well as using it in the SDP. If set, RTP packets will now
2230 originate from the media address instead of the operating system's "primary"
2235 * A new configuration section - ice_host_candidates - has been added to
2236 rtp.conf, allowing automatically discovered ICE host candidates to be
2237 overriden. This allows an Asterisk server behind a 1:1 NAT to send its
2238 external IP as a host candidate rather than relying on STUN to discover it.
2240 ------------------------------------------------------------------------------
2241 --- Functionality changes from Asterisk 13.6.0 to Asterisk 13.7.0 ------------
2242 ------------------------------------------------------------------------------
2246 * Added format attribute negotiation for the VP8 video codec. Format attribute
2247 negotiation is provided by the res_format_attr_vp8 module.
2251 * A new "timeout" user profile option has been added. This configures the number
2252 of seconds that a participant may stay in the ConfBridge after joining. When
2253 the time expires, the user is ejected from the conference and CONFBRIDGE_RESULT
2254 is set to "TIMEOUT" on the channel.
2258 * The websockets_enabled option has been added to the general section of
2259 sip.conf. The option is enabled by default to match the previous behavior.
2260 The option should be disabled when using res_pjsip_transport_websockets to
2261 ensure chan_sip will not conflict with PJSIP websockets.
2265 * The HOLD_INTERCEPT dialplan function now actually exists in the source tree.
2266 While support for the events was added in Asterisk 13.4.0, the function
2267 accidentally never made it in. That function is now present, and will cause
2268 the 'hold' raised by a channel to be intercepted and converted into an
2271 res_pjsip_outbound_registration
2272 -------------------------------
2273 * If res_statsd is loaded and a StatsD server is configured, basic statistics
2274 regarding the state of outbound registrations will now be emitted. This
2276 - A GAUGE statistic for the overall number of outbound registrations, i.e.:
2277 PJSIP.registrations.count
2278 - A GAUGE statistic for the overall number of outbound registrations in a
2279 particular state, e.g.:
2280 PJSIP.registrations.state.Registered
2284 * The ability to use "like" has been added to the pjsip list and show
2285 CLI commands. For instance: CLI> pjsip list endpoints like abc
2287 * If res_statsd is loaded and a StatsD server is configured, basic statistics
2288 regarding the state of PJSIP contacts will now be emitted. This includes:
2289 - A GAUGE statistic for the overall number of contacts in a particular
2291 PJSIP.contacts.states.Reachable
2292 - A TIMER statistic for the RTT time for each qualified contact, e.g.:
2293 PJSIP.contacts.alice@@127.0.0.1:5061.rtt
2295 res_sorcery_memory_cache
2296 ------------------------
2297 * A new caching strategy, full_backend_cache, has been added which caches
2298 all stored objects in the backend. When enabled all objects will be
2299 expired or go stale according to the configuration. As well when enabled
2300 all retrieval operations will be performed against the cache instead of
2305 * CALLERID(pres) is now documented as a valid alternative to setting both
2306 CALLERID(name-pres) and CALLERID(num-pres) at once. Some channel drivers,
2307 like chan_sip, don't make a distinction between the two: they take the
2308 least public value from name-pres and num-pres. By using CALLERID(pres)
2309 for reading and writing, you touch the same combined value in the dialplan.
2310 The same applies to CONNECTEDLINE(pres), REDIRECTING(orig-pres),
2311 REDIRECTING(to-pres) and REDIRECTING(from-pres).
2315 * A new module that emits StatsD statistics regarding Asterisk endpoints.
2316 This includes a total count of the number of endpoints, the count of the
2317 number of endpoints in the technology agnostic state of the endpoint -
2318 online or offline - as well as the number of channels associated with each
2319 endpoint. These are recorded as three different GAUGE statistics:
2321 - endpoints.state.{unknown|offline|online}
2322 - endpoints.{tech}.{resource}.channels
2325 ------------------------------------------------------------------------------
2326 --- Functionality changes from Asterisk 13.5.0 to Asterisk 13.6.0 ------------
2327 ------------------------------------------------------------------------------
2331 * The CHANNEL function, when used on a PJSIP channel, now exposes a 'call-id'
2332 extraction option when using with the 'pjsip' signalling option. It will
2333 return the SIP Call-ID associated with the INVITE request that established
2338 * Two new endpoint related events are now available: PeerStatusChange and
2339 ContactStatusChange. In particular, these events are useful when subscribing
2340 to all event sources, as they provide additional endpoint related
2341 information beyond the addition/removal of channels from an endpoint.
2343 * Added the ability to subscribe to all ARI events in Asterisk, regardless
2344 of whether the application 'controls' the resource. This is useful for
2345 scenarios where an ARI application merely wants to observe the system,
2346 as opposed to control it. There are two ways to accomplish this:
2347 (1) Via the WebSocket connection URI. A new query paramter, 'subscribeAll',
2348 has been added that, when present and True, will subscribe all
2349 specified applications to all ARI event sources in Asterisk.
2350 (2) Via the applications resource. An ARI client can, at any time, subscribe
2351 to all resources in an event source merely by not providing an explicit
2352 resource. For example, subscribing to an event source of 'channels:'
2353 as opposed to 'channels:12345' will subscribe the application to all
2356 ------------------------------------------------------------------------------
2357 --- Functionality changes from Asterisk 13.4.0 to Asterisk 13.5.0 ------------
2358 ------------------------------------------------------------------------------
2362 * A new ContactStatus event has been added that reflects res_pjsip contact
2363 lifecycle changes: Created, Removed, Reachable, Unreachable, Unknown.
2365 * Added the Linkedid header to the common channel headers listed for each
2366 channel in AMI events.
2370 * A new feature has been added that enables the retrieval of modules and
2371 module information through an HTTP request. Information on a single module
2372 can be also be retrieved. Individual modules can be loaded to Asterisk, as
2373 well as unloaded and reloaded.
2375 * A new resource has been added to the 'asterisk' resource, 'config/dynamic'.
2376 This resource allows for push configuration of sorcery derived objects
2377 within Asterisk. The resource supports creation, retrieval, updating, and
2378 deletion. Sorcery derived objects that are manipulated by this resource
2379 must have a sorcery wizard that supports the desired operations.
2381 * A new feature has been added that allows for the rotation of log channels
2382 through HTTP requests.
2387 * A new 'g726_non_standard' endpoint option has been added that, when set to
2388 'yes' and g.726 audio is negotiated, forces the codec to be treated as if it
2389 is AAL2 packed on the channel.
2391 * A new 'rtp_keepalive' endpoint option has been added. This option specifies
2392 an interval, in seconds, at which we will send RTP comfort noise packets to
2393 the endpoint. This functions identically to chan_sip's "rtpkeepalive" option.
2395 * New 'rtp_timeout' and 'rtp_timeout_hold' endpoint options have been added.
2396 These options specify the amount of time, in seconds, that Asterisk will wait
2397 before terminating the call due to lack of received RTP. These are identical
2398 to chan_sip's rtptimeout and rtpholdtimeout options.
2400 ------------------------------------------------------------------------------
2401 --- Functionality changes from Asterisk 13.3.0 to Asterisk 13.4.0 ------------
2402 ------------------------------------------------------------------------------
2406 * New 'rpid_immediate' option to control if connected line update information
2407 goes to the caller immediately or waits for another reason to send the
2408 connected line information update. See the online option documentation for
2409 more information. Defaults to 'no' as setting it to 'yes' can result in
2410 many unnecessary messages being sent to the caller.
2412 * The configuration setting 'progressinband' now defaults to 'no', which
2413 matches the actual behavior of previous versions.
2417 * A new CLI command has been added: "pjsip show settings", which shows
2418 both the global and system configuration settings.
2420 * A new aor option has been added: "qualify_timeout", which sets the timeout
2421 in seconds for a qualify. The default is 3 seconds. This overrides the
2422 hard coded 32 seconds in pjproject.
2424 * Endpoint status will now change to "Unreachable" when all contacts are
2425 unavailable. When any contact becomes available, the endpoint will status
2426 will change back to "Reachable".
2428 * A new global option has been added: "max_initial_qualify_time", which
2429 sets the maximum amount of time from startup that qualifies should be
2430 attempted on all contacts.
2434 * Two new events, 'ChannelHold' and 'ChannelUnhold', have been added to the
2435 events data model. These events are raised when a channel indicates a hold
2436 or unhold, respectively.
2440 * A new dialplan function, HOLD_INTERCEPT, has been added. This function, when
2441 placed on a channel, intercepts hold/unhold indications signalled by the
2442 channel and prevents them from moving on to other channels in a bridge with
2443 the hold initiator. Instead, AMI or ARI events are raised indicating that
2444 the channel wanted to place someone on hold. This allows external
2445 applications to implement their own custom hold/unhold logic.
2447 ------------------------------------------------------------------------------
2448 --- Functionality changes from Asterisk 13.2.0 to Asterisk 13.3.0 ------------
2449 ------------------------------------------------------------------------------
2451 chan_pjsip/app_transfer
2453 * The Transfer application, when used with chan_pjsip, now supports using
2454 a PJSIP endpoint as the transfer destination. This is in addition to
2455 explicitly specifying a SIP URI to transfer to.
2459 * The ARI /channels resource now supports a new operation, 'redirect'. The
2460 redirect operation will perform a technology and state specific redirection
2461 on the channel to a specified endpoint or destination. In the case of SIP
2462 technologies, this is either a 302 Redirect response to an on-going INVITE
2463 dialog or a SIP REFER request.
2467 * A new 'endpoint_identifier_order' option has been added that allows one to
2468 set the order by which endpoint identifiers are processed and checked. This
2469 option is specified under the 'global' type configuration section.
2471 ------------------------------------------------------------------------------
2472 --- Functionality changes from Asterisk 13.1.0 to Asterisk 13.2.0 ------------
2473 ------------------------------------------------------------------------------
2475 * New 'PJSIP_AOR' and 'PJSIP_CONTACT' dialplan functions have been added which
2476 allow examining PJSIP AORs or contacts from the dialplan.
2478 res_pjsip_outbound_registration
2480 * The 'pjsip send unregister' command now stops further registrations.
2482 * A new command 'pjsip send register' has been added which allows you to
2483 start or restart periodic registration. It can be used after a
2484 'send unregister' or after a 401 permanent error.
2486 res_pjsip_config_wizard
2488 * This is a new module that adds streamlined configuration capability for
2489 chan_pjsip. It's targeted at users who have lots of basic configuration
2490 scenarios like 'phone' or 'agent' or 'trunk'. Additional information
2491 can be found in the sample configuration file at
2492 config/samples/pjsip_wizard.conf.sample.
2496 * The T.38 negotiation timeout was previously hard coded at 5000 milliseconds
2497 and is now configurable via the 't38timeout' configuration option in
2498 res_fax.conf and via the fax options dialplan function 'FAXOPT(t38timeout)'.
2499 The default remains at 5000 milliseconds.
2503 * The ca_list_path transport parameter has been added for TLS transports. This
2504 option behaves similarly to the old sip.conf option "tlscapath". In order to
2505 use this, you must be using PJProject version 2.4 or higher.
2509 * The Originate operation now takes in an originator channel. The linked ID of
2510 this originator channel is applied to the newly originated outgoing channel.
2511 If using CEL this allows an association to be established between the two so
2512 it can be recognized that the originator is dialing the originated channel.
2514 * "language" (the default spoken language for the channel) is now included in
2515 the standard channel state output for suitable events.
2517 * The POST channels/{id} operation and the POST channels/{id}/continue operation
2518 now have a new "label" parameter. This allows for origination or continuation
2519 to a labeled priority in the dialplan instead of requiring a specific priority
2520 number. The ARI version has been bumped to 1.7.0 as a result.
2524 * "Language" (the default spoken language for the channel) is now included in
2525 the standard channel state output for suitable events.
2527 * AMI actions that return a list of events have been made to return consistent
2528 headers for the action response event starting the list and the list complete
2529 event. The AMI version has been bumped to 2.7.0 as a result.
2531 ------------------------------------------------------------------------------
2532 --- Functionality changes from Asterisk 13.0.0 to Asterisk 13.1.0 ------------
2533 ------------------------------------------------------------------------------
2537 * Event NewConnectedLine is emitted when the connected line information on
2542 * Event ChannelConnectedLine is emitted when the connected line information
2543 on a channel changes.
2548 The features.conf general section has three new configurable options:
2549 * transferdialattempts
2550 * transferretrysound
2551 * transferinvalidsound
2552 For more information on what these options do, see the Asterisk wiki:
2553 https://wiki.asterisk.org/wiki/x/W4fAAQ
2560 * New 'media_encryption_optimistic' endpoint setting. This will use SRTP
2561 when possible but does not consider lack of it a failure.
2563 res_pjsip_endpoint_identifer_ip
2565 * New CLI commands have been added: "pjsip show identif(y|ies)", which lists
2566 all configured PJSIP identify objects
2568 ------------------------------------------------------------------------------
2569 --- Functionality changes from Asterisk 12 to Asterisk 13 --------------------
2570 ------------------------------------------------------------------------------
2575 Asterisk 13 is the next Long Term Support (LTS) release of Asterisk. As such,
2576 the focus of development for this release of Asterisk was on improving the
2577 usability and features developed in the previous Standard release, Asterisk 12.
2578 Beyond a general refinement of end user features, development focussed heavily
2579 on the Asterisk APIs - the Asterisk Manager Interface (AMI) and the Asterisk
2580 REST Interface (ARI) - and the PJSIP stack in Asterisk. Some highlights of the
2581 new features include:
2583 * Asterisk security events are now provided via AMI, allowing end users to
2584 monitor their Asterisk system in real time for security related issues.
2585 * External control of Message Waiting Indicators (MWI) through both AMI and ARI.
2586 * Reception/transmission of out of call text messages using any supported
2587 channel driver/protocol stack through ARI.
2588 * Resource List Server support in the PJSIP stack, providing subscriptions to
2589 lists of resources and batched delivery of NOTIFY requests.
2590 * Inter-Asterisk distributed device state and mailbox state using the PJSIP
2593 It is important to note that Asterisk 13 is built on the architecture developed
2594 during the previous Standard release, Asterisk 12. Users upgrading to
2595 Asterisk 13 should read about the new features in Asterisk 12 later in this file
2596 (see Functionality changes from Asterisk 11 to Asterisk 12), as well as the
2597 UPGRADE-12.txt delivered with this release. In particular, users upgrading to
2598 Asterisk 13 from a release prior to Asterisk 12 should read the specifications
2599 on AMI, CDRs, and CEL on the Asterisk wiki:
2600 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
2601 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
2602 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
2604 Many new featuers in Asterisk 13 were introduced in point releases of
2605 Asterisk 12. Following this section - which documents the changes from all
2606 versions of Asterisk 12 to Asterisk 13 - users should examine the new features
2607 that were introduced in the point releases of Asterisk 12, as they are also
2608 included in Asterisk 13.
2610 Finally, all users upgrading to Asterisk 13 should read the UPGRADE.txt file
2611 delivered with this release.
2616 * Sample config files have been moved from configs/ to a sub-folder of that
2619 * The menuselect utility has been pulled into the Asterisk repository. As a
2620 result, the libxml2 development library is now a required dependency for
2623 * A new Compiler Flag, REF_DEBUG, has been added. When enabled, reference
2624 counted objects will emit additional debug information to the refs log file
2625 located in the standard Asterisk log file directory. This log file is useful
2626 in tracking down object leaks and other reference counting issues. Prior to
2627 this version, this option was only available by modifying the source code
2628 directly. This change also includes a new script, refcounter.py, in the
2629 contrib folder that will process the refs log file. Note that this replaces
2630 the refcounter utility that could be built from the utils directory.
2638 * This module was deprecated and has been removed. Users of app_dahdibarge
2639 should use ChanSpy instead.
2643 * New options to play a beep when starting a recording and stopping a recording
2644 have been added. The option "p" will play a beep to the channel that starts
2645 the recording. The option "P" will play a beep to the channel that stops the
2650 * Queue rules can now be stored in a database table, queue_rules. Unlike other
2651 RealTime tables, the queue_rules table is only examined on module load or
2652 module reload. A new general setting has been added to queuerules.conf,
2653 'realtime_rules', which, when set to 'yes', will cause app_queue to look in
2654 RealTime for additional queue rules to parse. Note that both the file and
2655 the database can be used as a provide of queue rules when 'realtime_rules'
2658 When app_queue is reloaded, all rules are re-parsed and loaded into memory.
2659 There is no caching of RealTime queue rules.
2663 * This module was deprecated and has been removed. Users of app_readfile
2664 should use func_env's FILE function instead.
2668 * The 'say' family of dialplan applications now support the Japanese
2669 language. The 'language' parameter in say.conf now recognizes a setting of
2670 'ja', which will enable Japanese language specific mechanisms for playing
2671 back numbers, dates, and other items.
2672 * Counting, enumeration and dates now supports Icelandic grammar with the
2673 'language' parameter set to 'is'.
2677 * This module was deprecated and has been removed. Users of app_saycountpl
2678 should use the Say family of applications.
2682 * The SetMusicOnHold dialplan application was deprecated and has been removed.
2683 Users of the application should use the CHANNEL function's musicclass
2688 * The WaitMusicOnHold dialplan application was deprecated and has been
2689 removed. Users of the application should use MusicOnHold with a duration
2694 * VoiceMail and VoiceMailMain now support the Japanese language. The
2695 'language' parameter in voicemail.conf now recognizes a setting of 'ja',
2696 which will enable prompts to be played back using a Japanese grammatical
2697 structure. Additional prompts are necessary for this functionality,
2699 - jb-arimasu: there is
2700 - jb-arimasen: there is not
2701 - jb-oshitekudasai: please press
2707 * Add the ability to specify multiple email addresses in configuration,
2716 * This module was deprecated and has been removed. Users of cdr_sqlite
2717 should use cdr_sqlite3_custom.
2721 * Added the ability to support PostgreSQL application_name on connections.
2722 This allows PostgreSQL to display the configured name in the
2723 pg_stat_activity view and CSV log entries. This setting is configurable
2724 for cdr_pgsql via the appname configuration setting in cdr_pgsql.conf.
2732 * Added the ability to support PostgreSQL application_name on connections.
2733 This allows PostgreSQL to display the configured name in the
2734 pg_stat_activity view and CSV log entries. This setting is configurable
2735 for cel_pgsql via the appname configuration setting in cel_pgsql.conf.
2743 * SS7 support now requires libss7 v2.0 or later.
2745 * Added SS7 support for connected line and redirecting.
2747 * Most SS7 CLI commands are reworked as well as new SS7 commands added.
2748 See online CLI help.
2750 * Added several SS7 config option parameters described in
2751 chan_dahdi.conf.sample.
2755 * This module was deprecated and has been removed. Users of chan_gtalk
2756 should use chan_motif.
2760 * This module was deprecated and has been removed. Users of chan_h323
2761 should use chan_ooh323.
2765 * This module was deprecated and has been removed. Users of chan_jingle
2766 should use chan_motif.
2770 * Added the CLI command 'pjsip list ciphers' so a user can know what
2771 OpenSSL names are available on their system for the pjsip.conf cipher
2776 * The SIPPEER dialplan function no longer supports using a colon as a
2777 delimiter for parameters. The parameters for the function should be
2778 delimited using a comma.
2780 * The SIPCHANINFO dialplan function was deprecated and has been removed. Users
2781 of the function should use the CHANNEL function instead.
2789 * Added functional peeraccount support. Except for Queue, the
2790 accountcode propagation is now consistently propagated to outgoing
2791 channels before dialing. The channel accountcode can change from its
2792 original non-empty value on channel creation for the following specific
2793 reasons. One, dialplan sets it using CHANNEL(accountcode). Two, an
2794 originate method that can specify an accountcode value. Three, the
2795 calling channel propagates its peeraccount or accountcode to the
2796 outgoing channel's accountcode before dialing. The change has two
2797 visible effects. One, local channels now cross accountcode and
2798 peeraccount across the special bridge between the ;1 and ;2 channels
2799 just like channels between normal bridges. Two, the
2800 CHANNEL(peeraccount) value can now be set before Dial and FollowMe to
2801 set the accountcode on the outgoing channel(s).
2803 For Queue, an outgoing channel's non-empty accountcode will not change
2804 unless explicitly set by CHANNEL(accountcode). The change has three
2805 visible effects. One, local channels now cross accountcode and
2806 peeraccount across the special bridge between the ;1 and ;2 channels
2807 just like channels between normal bridges. Two, the queue member will
2808 get an accountcode if it doesn't have one and one is available from the
2809 calling channel's peeraccount. Three, accountcode propagation includes
2810 local channel members where the accountcodes are propagated early
2811 enough to be available on the ;2 channel.
2815 * New DeviceStateChanged and PresenceStateChanged AMI events have been added.
2816 These events are emitted whenever a device state or presence state change
2817 occurs. The events are controlled by res_manager_device_state.so and
2818 res_manager_presence_state.so. If the high frequency of these events is
2819 problematic for you, do not load these modules.
2821 * Added DialplanExtensionAdd and DialplanExtensionRemove AMI commands. They
2822 work in basically the same way as the 'dialplan add extension' and
2823 'dialplan remove extension' CLI commands respectively.
2825 * New AMI action LoggerRotate reloads and rotates logger in the same manner
2826 as CLI command 'logger rotate'
2828 * New AMI Actions FAXSessions, FAXSession, and FAXStats replicate the
2829 functionality of CLI commands 'fax show sessions', 'fax show session',
2830 and fax show stats' respectively.
2832 * New AMI actions PRIDebugSet, PRIDebugFileSet, and PRIDebugFileUnset
2833 enable manager control over PRI debugging levels and file output.
2835 * AMI action PJSIPNotify may now send to a URI instead of only to a PJSIP
2836 endpoint as long as a default outbound endpoint is set. This also applies
2837 to the equivalent CLI command (pjsip send notify)
2839 * The AMI action PJSIPShowEndpoint now includes ContactStatusDetail sections
2840 that give information on Asterisk's attempts to qualify the endpoint.
2842 * The DialEnd event will now contain a Forward header if the dial is ending
2843 due to the call being forwarded. The contents of the Forward header is the
2844 extension in the number to which the call is being forwarded.
2848 * The "bridge_technology" extra field key has been added to BRIDGE_ENTER
2849 and BRIDGE_EXIT events.
2853 * Channel variables are now substituted in arguments passed to applications
2854 run by using dynamic features.
2858 * The TLS core in Asterisk now supports Perfect Forward Secrecy (PFS).
2859 Enabling PFS is attempted by default, and is dependent on the configuration
2860 of the module using TLS.
2861 - Ephemeral ECDH (ECDHE) is enabled by default. To disable it, do not
2862 specify a ECDHE cipher suite in sip.conf, for example:
2863 tlscipher=AES128-SHA:DES-CBC3-SHA
2864 - Ephemeral DH (DHE) is disabled by default. To enable it, add DH parameters
2865 into the private key file, e.g., sip.conf tlsprivatekey. For example, the
2866 default dh2048.pem - see
2867 http://www.opensource.apple.com/source/OpenSSL098/OpenSSL098-35.1/src/apps/dh2048.pem?txt
2868 - Because clients expect the server to prefer PFS, and because OpenSSL sorts
2869 its cipher suites by bit strength, see "openssl ciphers -v DEFAULT".
2870 Consider re-ordering your cipher suites in the respective configuration
2872 tlscipher=AES128+kEECDH:AES128+kEDH:3DES+kEDH:AES128-SHA:DES-CBC3-SHA:-ADH:-AECDH
2873 will use PFS when offered by the client. Clients which do not offer PFS
2874 fall-back to AES-128 (or even 3DES, as recommended by RFC 3261).
2882 * The JACK_HOOK function now supports audio with a sample rate higher than
2891 * Added the ability to support PostgreSQL application_name on connections.
2892 This allows PostgreSQL to display the configured name in the
2893 pg_stat_activity view and CSV log entries. This setting is configurable
2894 for res_config_pgsql via the dbappname configuration setting in
2897 res_pjsip_outbound_publish
2899 * A new module, res_pjsip_outbound_publish provides the mechanisms for sending
2900 PUBLISH requests for specific event packages to another SIP User Agent.
2904 * The publish/subscribe core module has been updated to support RFC 4662
2905 Resource Lists, allowing Asterisk to act as a Resource List Server (RLS).
2906 Resource lists are configured in pjsip.conf under a new object type,
2907 resource_list. Resource lists can contain either message-summary or presence
2908 events, and can be composed of specific resources that provide the event or
2909 other resource lists.
2911 * Inbound publication support is provided by a new object, inbound-publication.
2912 This configures res_pjsip_pubsub to accept PUBLISH requests from a particular
2913 resource. Which events are accepted is constructed dynamically; see
2914 res_pjsip_publish_asterisk for more information.
2916 res_pjsip_publish_asterisk
2918 * A new module, res_pjsip_publish_asterisk adds support for PUBLISH requests of
2919 Asterisk information to other Asterisk servers. This module is intended only
2920 for Asterisk to Asterisk exchanges of information. Currently, this includes
2921 both mailbox state and device state information.
2923 ------------------------------------------------------------------------------
2924 --- Functionality changes from Asterisk 12.4.0 to Asterisk 12.5.0 ------------
2925 ------------------------------------------------------------------------------
2929 * Stored recordings now support a new operation, copy. This will take an
2930 existing stored recording and copy it to a new location in the recordings
2933 * LiveRecording objects now have three additional fields that can be reported
2934 in a RecordingFinished ARI event:
2935 - total_duration: the duration of the recording
2936 - talking_duration: optional. The duration of talking detected in the
2937 recording. This is only available if max_silence_seconds was specified
2938 when the recording was started.
2939 - silence_duration: optional. The duration of silence detected in the
2940 recording. This is only available if max_silence_seconds was specified
2941 when the recording was started.
2942 Note that all duration values are reported in seconds.
2944 * Users of ARI can now send and receive out of call text messages. Messages
2945 can be sent directly to a particular endpoint, or can be sent to the
2946 endpoints resource directly and inferred from the URI scheme. Text
2947 messages are passed to ARI clients as TextMessageReceived events. ARI
2948 clients can choose to receive text messages by subscribing to the particular
2949 endpoint technology or endpoints that they are interested in.
2951 * The applications resource now supports subscriptions to all endpoints of
2952 a particular channel technology. For example, subscribing to an eventSource
2953 of 'endpoint:PJSIP' will subscribe to all PJSIP endpoints.
2957 * The endpoint configuration object now supports 'accountcode'. Any channel
2958 created for an endpoint with this setting will have its accountcode set
2959 to the specified value.
2963 * A new module, res_hep_rtcp, has been added that will forward RTCP call
2964 statistics to a HEP capture server. See res_hep for more information.
2968 * Function AUDIOHOOK_INHERIT has been deprecated. Audiohooks are now
2969 unconditionally inhereted through masquerades. As a side benefit, more
2970 than one audiohook of a given type may persist through a masquerade now.
2972 ------------------------------------------------------------------------------
2973 --- Functionality changes from Asterisk 12.3.0 to Asterisk 12.4.0 ------------
2974 ------------------------------------------------------------------------------
2978 * Returns new AGENT_STATUS value "NOT_CONNECTED" if the agent fails to
2979 connect with an incoming caller after being alerted to the presence
2980 of the incoming caller. The most likely reason this would happen is
2981 the agent did not acknowledge the call in time.
2985 * New events have been added for the TALK_DETECT function. When the function
2986 is used on a channel, ChannelTalkingStart/ChannelTalkingStop events will be
2987 emitted to connected AMI clients indicating the start/stop of talking on
2992 * New event models have been aded for the TALK_DETECT function. When the
2993 function is used on a channel, ChannelTalkingStarted/ChannelTalkingFinished
2994 events will be emitted to connected WebSockets subscribed to the channel,
2995 indicating the start/stop of talking on the channel.
2999 * A new function, TALK_DETECT, has been added. When set on a channel, this
3000 fucntion causes events indicating the starting/stoping of talking on said
3001 channel to be emitted to both AMI and ARI clients.
3003 ------------------------------------------------------------------------------
3004 --- Functionality changes from Asterisk 12.2.0 to Asterisk 12.3.0 ------------
3005 ------------------------------------------------------------------------------
3009 * A new Playback URI 'tone' has been added. Tones are specified either as
3010 an indication name (e.g. 'tone:busy') from indications.conf or as a tone
3011 pattern (e.g. 'tone:240/250,0/250'). Tones differ from normal playback
3012 URIs in that they must be stopped manually and will continue to occupy
3013 a channel's ARI control queue until they are stopped. They also can not
3014 be rewound or fastforwarded.
3016 * User events can now be generated from ARI. Events can be signalled with
3017 arbitrary json variables, and include one or more of channel, bridge, or
3018 endpoint snapshots. An application must be specified which will receive
3019 the event message (other applications can subscribe to it). The message
3020 will also be delivered via AMI provided a channel is attached. Dialplan
3021 generated user event messages are still transmitted via the channel, and
3022 will only be received by a stasis application they are attached to or if
3023 the channel is subscribed to.
3027 * SIP peers can now specify 'trust_id_outbound' which affects RPID/PAI
3028 fields for prohibited callingpres information. Values are legacy, no, and
3029 yes. By default, legacy is used.
3030 trust_id_outbound=legacy - behavior remains the same as 1.8.26.1. When
3031 dealing with prohibited callingpres and sendrpid=pai/rpid, RPID/PAI
3032 headers are appended to outbound SIP messages just as they are with
3033 allowed callingpres values, but data about the remote party's identity is
3035 When sendrpid=rpid, only the remote party's domain is anonymized.
3036 trust_id_outbound=no - when dealing with prohibited callingpres, RPID/PAI
3037 headers are not sent.
3038 trust_id_outbound=yes - RPID/PAI headers are applied with the full remote
3039 party information in tact even for prohibited callingpres information.
3040 In the case of PAI, a Privacy: id header will be appended for prohibited
3041 calling information to communicate that the private information should
3042 not be relayed to untrusted parties.
3046 * Manager action 'Park' now takes an additional argument 'AnnounceChannel'
3047 which can be used to announce the parked call's location to an arbitrary
3048 channel in a bridge. If 'Channel' and 'TimeoutChannel' are now the two
3049 parties in a one to one bridge, 'TimeoutChannel' is treated as having
3050 parked 'Channel' like with the Park Call DTMF feature and will receive
3051 announcements prior to being hung up.
3053 ------------------------------------------------------------------------------
3054 --- Functionality changes from Asterisk 12.1.0 to Asterisk 12.2.0 ------------
3055 ------------------------------------------------------------------------------
3059 * Record application now has an option 'o' which allows 0 to act as an exit
3060 key setting the RECORD_STATUS variable to 'OPERATOR' instead of 'DTMF'
3063 --------------------------
3064 * ChanSpy now accepts a channel uniqueid or a fully specified channel name
3065 as the chanprefix parameter if the 'u' option is specified.
3068 --------------------------
3069 * CONFBRIDGE dialplan function is now capable of creating/modifying dynamic
3070 conference user menus.
3072 * CONFBRIDGE dialplan function is now capable of removing dynamic conference
3073 menus, bridge settings, and user settings that have been applied by the
3074 CONFBRIDGE dialplan function.
3076 * The ConfBridge dialplan application now sets a channel variable,
3077 CONFBRIGE_RESULT, upon exiting. This variable can be used to determine
3078 how a channel exited the conference.
3080 * Added conference user option 'announce_join_leave_review'. This option
3081 implies 'announce_join_leave' with the added effect that the user will
3082 be asked if they want to confirm or re-record the recording of their
3083 name when entering the conference
3086 --------------------------
3087 * At exit, the Directory application now sets a channel variable
3088 DIRECTORY_RESULT to one of the following based on the reason for exiting:
3089 OPERATOR user requested operator by pressing '0' for operator
3090 ASSISTANT user requested assistant by pressing '*' for assistant
3091 TIMEOUT user pressed nothing and Directory stopped waiting
3092 HANGUP user's channel hung up
3093 SELECTED user selected a user from the directory and is routed
3094 USEREXIT user pressed '#' from the selection prompt to exit
3095 FAILED directory failed in a way that wasn't accounted for. Dang.
3099 * Monitor() - A new option, B(), has been added that will turn on a periodic
3100 beep while the call is being recorded.
3103 --------------------------
3104 * MusicOnHold streams (all modes other than "files") now support wide band
3108 --------------------------
3109 * Added options 'b' and 'B' to apply predial handlers for outgoing calls
3110 and for the channel executing Page respectively.
3113 --------------------------
3114 * PickupChan now accepts channel uniqueids of channels to pickup.
3117 --------------------------
3118 * If a channel variable SAY_DTMF_INTERRUPT is present on a channel and set
3119 to 'true' (case insensitive), then any Say application (SayNumber,
3120 SayDigits, SayAlpha, SayAlphaCase, SayUnixTime, and SayCounted) will
3121 anticipate DTMF. If DTMF is received, these applications will behave like
3122 the background application and jump to the received extension once a match
3123 is established or after a short period of inactivity.
3126 -------------------------
3127 * A new function, MIXMONITOR, has been added to allow access to individual
3128 instances of MixMonitor on a channel.
3130 * A new option, B(), has been added that will turn on a periodic beep while the
3131 call is being recorded.
3135 -------------------------
3138 -------------------------
3139 * TEL URI support for inbound INVITE requests has been added. chan_sip will
3140 now handle TEL schemes in the Request and From URIs. The phone-context in
3141 the Request URI will be stored in the SIPURIPHONECONTEXT channel variable on
3142 the inbound channel.
3146 * Exposed sorcery-based configuration files like pjsip.conf to dialplans via
3147 the new AST_SORCERY diaplan function.
3149 * Core Show Locks output now includes Thread/LWP ID if the platform
3150 supports this feature.
3152 * New "logger add channel" and "logger remove channel" CLI commands have
3153 been added to allow creation and deletion of dynamic logger channels
3154 without configuration changes. These dynamic logger channels will only
3155 exist until the next restart of asterisk.
3159 * The live recording object on recording events now contains a target_uri
3160 field which contains the URI of what is being recorded.
3162 * The bridge type used when creating a bridge is now a comma separated list of
3163 bridge properties. Valid options are: mixing, holding, dtmf_events, and
3166 * A channelId can now be provided when creating a channel, either in the
3167 uri (POST channels/my-channel-id) or as query parameter. A local channel
3168 will suffix the second channel id with ';2' unless provided as query
3169 parameter otherChannelId.
3171 * A bridgeId can now be provided when creating a bridge, either in the uri
3172 (POST bridges/my-bridge-id) or as a query parameter.
3174 * A playbackId can be provided when starting a playback, either in the uri
3175 (POST channels/my-channel-id/play/my-playback-id /
3176 POST bridges/my-bridge-id/play/my-playback-id) or as a query parameter.
3178 * A snoop channel can be started with a snoopId, in the uri or query.
3182 * Originate now takes optional parameters ChannelId and OtherChannelId,
3183 used to set the UniqueId on creation. The other id is assigned to the
3184 second channel when dialing LOCAL, or defaults to appending ;2 if only
3185 the single Id is given.
3187 * The Mixmonitor action now has a "Command" header that can be used to
3188 indicate a post-process command to run once recording finishes.
3192 * A new set of Alembic scripts has been added for CDR tables. This will create
3193 a 'cdr' table with the default schema that Asterisk expects.
3198 * A new function was added: PERIODIC_HOOK. This allows running a periodic
3199 dialplan hook on a channel. Any audio generated by this hook will be
3200 injected into the call.
3208 * A new module, res_hep, has been added, that acts as a generic packet
3209 capture agent for the Homer Encapsulation Protocol (HEP) version 3.
3210 It can be configured via hep.conf. Other modules can use res_hep to send
3211 message traffic to a HEP capture server.
3215 * A new module, res_hep_pjsip, has been added that will forward PJSIP
3216 message traffic to a HEP capture server. See res_hep for more
3221 * transport and endpoint ToS options (tos, tos_audio, and tos_video) may now
3222 be set as the named set of ToS values (cs0-cs7, af11-af43, ef).
3224 * Added the following new CLI commands:
3225 - "pjsip show contacts" - list all current PJSIP contacts.
3226 - "pjsip show contact" - show specific information about a current PJSIP
3228 - "pjsip show channel" - show detailed information about a PJSIP channel.
3230 res_pjsip_multihomed
3232 * A new module, res_pjsip_multihomed handles situations where the system
3233 Asterisk is running out has multiple interfaces. res_pjsip_multihomed
3234 determines which interface should be used during message sending.
3236 res_pjsip_pidf_digium_body_supplement
3238 * A new module, res_pjsip_pidf_digium_body_supplement provides NOTIFY
3239 request body formatting for presence support in Digium phones.
3241 res_pjsip_send_to_voicemail
3243 * A new module, res_pjsip_send_to_voicemail allows for REFER requests with
3244 particular headers to transfer a PJSIP channel directly to a particular
3245 extension that has VoiceMail. This is intended to be used with Digium
3246 phones that support this feature.
3248 res_pjsip_outbound_registration
3250 * A new CLI command has been added: "pjsip show registrations", which lists
3251 all configured PJSIP registrations
3254 ------------------------------------------------------------------------------
3255 --- Functionality changes from Asterisk 12.0.0 to Asterisk 12.1.0 ------------
3256 ------------------------------------------------------------------------------
3260 * Added a new module that provides AMI control over MWI within Asterisk,
3261 res_mwi_external_ami. Note that this module depends on res_mwi_external;
3262 for more information on enabling this module, see res_mwi_external.
3263 This module provides the MWIGet/MWIUpdate/MWIDelete actions, as well as
3264 the MWIGet/MWIGetComplete events.
3266 * The DialStatus field in the DialEnd event can now contain additional
3267 statuses that convey how the dial operation terminated. This includes
3268 ABORT, CONTINUE, and GOTO.
3270 * AMI will now emit security events. A new class authorization has been
3271 added in manager.conf for the security events, 'security'. The new events
3273 - FailedACL - raised when a request violates an ACL check
3274 - InvalidAccountID - raised when a request fails an authentication
3275 check due to an invalid account ID
3276 - SessionLimit - raised when a request fails due to exceeding the
3277 number of allowed concurrent sessions for a service
3278 - MemoryLimit - raised when a request fails due to an internal memory
3280 - LoadAverageLimit - raised when a request fails because a configured
3281 load average limit has been reached
3282 - RequestNotAllowed - raised when a request is not allowed by
3284 - AuthMethodNotAllowed - raised when a request used an authentication
3285 method not allowed by the service
3286 - RequestBadFormat - raised when a request is received with bad formatting
3287 - SuccessfulAuth - raised when a request successfully authenticates
3288 - UnexpectedAddress - raised when a request has a different source address
3289 then what is expected for a session already in progress with a service
3290 - ChallengeResponseFailed - raised when a request's attempt to authenticate
3291 has been challenged, and the request failed the authentication challenge
3292 - InvalidPassword - raised when a request provides an invalid password
3293 during an authentication attempt
3294 - ChallengeSent - raised when an Asterisk service send an authentication
3295 challenge to a request
3296 - InvalidTransport - raised when a request attempts to use a transport not
3297 allowed by the Asterisk service
3299 * Bridge related events now have two additional fields: BridgeName and
3300 BridgeCreator. BridgeName is a descriptive name for the bridge;
3301 BridgeCreator is the name of the entity that created the bridge. This
3302 affects the following events: ConfbridgeStart, ConfbridgeEnd,
3303 ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord,
3304 ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer,
3305 AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave
3309 * The Bridge data model now contains the additional fields 'name' and
3310 'creator'. The 'name' field conveys a descriptive name for the bridge;
3311 the 'creator' field conveys the name of the entity that created the bridge.
3312 This affects all responses to HTTP requests that return a Bridge data model
3313 as well as all event derived data models that contain a Bridge data model.
3314 The POST /bridges operation may now optionally specify a name to give to
3315 the bridge being created.
3317 * Added a new ARI resource 'mailboxes' which allows the creation and
3318 modification of mailboxes managed by external MWI. Modules res_mwi_external
3319 and res_stasis_mailbox must be enabled to use this resource. For more
3320 information on external MWI control, see res_mwi_external.
3322 * Added new events for externally initiated transfers. The event
3323 BridgeBlindTransfer is now raised when a channel initiates a blind transfer
3324 of a bridge in the ARI controlled application to the dialplan; the
3325 BridgeAttendedTransfer event is raised when a channel initiates an
3326 attended transfer of a bridge in the ARI controlled application to the
3329 * Channel variables may now be specified as a body parameter to the
3330 POST /channels operation. The 'variables' key in the JSON is interpreted
3331 as a sequence of key/value pairs that will be added to the created channel
3332 as channel variables. Other parameters in the JSON body are treated as
3333 query parameters of the same name.
3337 * Asterisk's HTTP server now supports chunked Transfer-Encoding. This will be
3338 automatically handled by the HTTP server if a request is received with a
3339 Transfer-Encoding type of "chunked".
3343 * Path support has been added with the 'support_path' option in registration
3346 * A 'debug' option has been added to the globals section that will allow
3347 sip messages to be logged.
3349 * A 'set_var' option has been added to endpoints that will automatically
3350 set the desired variable(s) on a channel created for that endpoint.
3352 * Several new tables and columns have been added to the realtime schema for
3353 the res_pjsip related modules. See the UPGRADE.txt notes for updating
3354 the database schema.
3358 * A new module, res_mwi_external, has been added to Asterisk. This module
3359 acts as a base framework that other modules can build on top of to allow
3360 an external system to control MWI within Asterisk. For implementations
3361 that make use of res_mwi_external, see res_mwi_external_ami and
3362 res_ari_mailboxes. Note that res_mwi_external canflicts with other modules
3363 that may produce MWI themselves, such as app_voicemail. res_mwi_external
3364 and other modules that depend on it cannot be built or loaded with
3365 app_voicemail present.
3369 * DNS functionality will now automatically be enabled if the system configured
3370 nameservers can be retrieved. If the system configured nameservers can not be
3371 retrieved the functionality will resort to using system resolution. Functionalty
3372 such as SRV records and failover will not be available if system resolution
3375 ------------------------------------------------------------------------------
3376 --- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
3377 ------------------------------------------------------------------------------
3382 Asterisk 12 is a standard release of the Asterisk project. As such, the
3383 focus of development for this release was on core architectural changes and
3384 major new features. This includes:
3385 * A more flexible bridging core based on the Bridging API
3386 * A new internal message bus, Stasis
3387 * Major standardization and consistency improvements to AMI
3388 * Addition of the Asterisk RESTful Interface (ARI)
3389 * A new SIP channel driver, chan_pjsip
3390 In addition, as the vast majority of bridging in Asterisk was migrated to the
3391 Bridging API used by ConfBridge, major changes were made to most of the
3392 interfaces in Asterisk. This includes not only AMI, but also CDRs and CEL.
3394 Specifications have been written for the affected interfaces. These
3395 specifications are available on the Asterisk wiki:
3396 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
3397 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
3398 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
3400 It is *highly* recommended that anyone migrating to Asterisk 12 read the
3401 information regarding its release both in this file and in the accompanying
3402 UPGRADE.txt file. More detailed information on the major changes can be found
3403 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/0YCLAQ.
3408 * Added build option DISABLE_INLINE. This option can be used to work around a
3409 bug in gcc. For more information, see
3410 http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816
3412 * Removed the CHANNEL_TRACE development mode build option. Certain aspects of
3413 the CHANNEL_TRACE build option were incompatible with the new bridging
3416 * Asterisk now optionally uses libxslt to improve XML documentation generation
3417 and maintainability. If libxslt is not available on the system, some XML
3418 documentation will be incomplete.
3420 * Asterisk now depends on libjansson. If a package of libjansson is not
3421 available on your distro, please see http://www.digip.org/jansson/.
3423 * Asterisk now depends on libuuid and, optionally, uriparser. It is
3424 recommended that you install uriparser, even if it is optional.
3426 * The new SIP stack and channel driver uses a particular version of PJSIP.
3427 Please see https://wiki.asterisk.org/wiki/x/J4GLAQ for more information on
3428 configuring and installing PJSIP for usage with Asterisk.
3430 * Optional API was re-implemented to be more portable, and no longer requires
3431 weak reference support from the compiler. The build option OPTIONAL_API may
3432 be disabled to disable Optional API support.
3439 * Along with AgentRequest, this application has been modified to be a
3440 replacement for chan_agent. The act of a channel calling the AgentLogin
3441 application places the channel into a pool of agents that can be
3442 requested by the AgentRequest application. Note that this application, as
3443 well as all other agent related functionality, is now provided by the
3444 app_agent_pool module. See chan_agent and AgentRequest for more information.
3446 * This application no longer performs agent authentication. If authentication
3447 is desired, the dialplan needs to perform this function using the
3448 Authenticate or VMAuthenticate application or through an AGI script before
3451 * If this application is called and the agent is already logged in, the
3452 dialplan will continue exection with the AGENT_STATUS channel variable set
3453 to ALREADY_LOGGED_IN.
3455 * The agents.conf schema has changed. Rather than specifying agents on a
3456 single line in comma delineated fashion, each agent is defined in a separate
3457 context. This allows agents to use the power of context templates in their
3460 * A number of parameters from agents.conf have been removed. This includes
3461 maxloginretries, autologoffunavail, updatecdr, goodbye, group, recordformat,
3462 urlprefix, and savecallsin. These options were obsoleted by the move from
3463 a channel driver model to the bridging/application model provided by
3468 * A new application, this will request a logged in agent from the pool and
3469 bridge the requested channel with the channel calling this application.
3470 Logged in agents are those channels that called the AgentLogin application.
3471 If an agent cannot be requested from the pool, the AGENT_STATUS dialplan
3472 application will be set with an appropriate error value.
3474 AgentMonitorOutgoing
3476 * This application has been removed. It was a holdover from when
3477 AgentCallbackLogin was removed.
3481 * Added support for additional Ademco DTMF signalling formats, including
3482 Express 4+1, Express 4+2, High Speed and Super Fast.
3484 * Added channel variable ALARMRECEIVER_CALL_LIMIT. This sets the maximum
3485 call time, in milliseconds, to run the application.
3487 * Added channel variable ALARMRECEIVER_RETRIES_LIMIT. This sets the
3488 maximum number of times to retry the call.
3490 * Added a new configuration option answait. If set, the AlarmReceiver
3491 application will wait the number of milliseconds specified by answait
3492 after the channel has answered. Valid values range between 500
3493 milliseconds and 10000 milliseconds.
3495 * Added configuration option no_group_meta. If enabled, grouping of metadata
3496 information in the AlarmReceiver log file will be skipped.
3500 * It is now no longer possible to bypass updating the CDR on the channel
3501 when answering. CDRs reflect the state of the channel and will always
3502 reflect the time they were Answered.
3506 * A new application in Asterisk, this will place the calling channel
3507 into a holding bridge, optionally entertaining them with some form of
3508 media. Channels participating in a holding bridge do not interact with
3509 other channels in the same holding bridge. Optionally, however, a channel
3510 may join as an announcer. Any media passed from an announcer channel is
3511 played to all channels in the holding bridge. Channels leave a holding
3512 bridge either when an optional timer expires, or via the ChannelRedirect
3513 application or AMI Redirect action.
3517 * All participants in a bridge can now be kicked out of a conference room
3518 by specifying the channel parameter as 'all' in the ConfBridge kick CLI
3519 command, i.e., 'confbridge kick <conference> all'
3521 * CLI output for the 'confbridge list' command has been improved. When
3522 displaying information about a particular bridge, flags will now be shown
3523 for the participating users indicating properties of that user.
3525 * The ConfbridgeList event now contains the following fields: WaitMarked,
3526 EndMarked, and Waiting. This displays additional properties about the
3527 user's profile, as well as whether or not the user is waiting for a
3528 Marked user to enter the conference.
3530 * Added a new option for conference recording, record_file_append. If enabled,
3531 when the recording is stopped and then re-started, the existing recording
3532 will be used and appended to.
3534 * ConfBridge now has the ability to set the language of announcements to the
3535 conference. The language can be set on a bridge profile in confbridge.conf
3536 or by the dialplan function CONFBRIDGE(bridge,language)=en.
3540 * The channel variable CPLAYBACKSTATUS may now return the value
3541 'REMOTESTOPPED'. This occurs when playback is stopped by a remote interface,
3542 such as AMI. See the AMI action ControlPlayback for more information.
3546 * Added the 'a' option, which allows the caller to enter in an additional
3547 alias for the user in the directory. This option must be used in conjunction
3548 with the 'f', 'l', or 'b' options. Note that the alias for a user can be
3549 specified in voicemail.conf.
3553 * The output of DumpChan no longer includes the DirectBridge or IndirectBridge
3554 fields. Instead, if a channel is in a bridge, it includes a BridgeID field
3555 containing the unique ID of the bridge that the channel happens to be in.
3559 * ForkCDR no longer automatically resets the forked CDR. See the 'r' option
3560 for more information.
3562 * Variables are no longer purged from the original CDR. See the 'v' option for
3565 * The 'A' option has been removed. The Answer time on a CDR is never updated
3568 * The 'd' option has been removed. The disposition on a CDR is a function of
3569 the state of the channel and cannot be altered.
3571 * The 'D' option has been removed. Who the Party B is on a CDR is a function
3572 of the state of the respective channels involved in the CDR and cannot be
3575 * The 'r' option has been changed. Previously, ForkCDR always reset the CDR
3576 such that the start time and, if applicable, the answer time was updated.
3577 Now, by default, ForkCDR simply forks the CDR, maintaining any times. The
3578 'r' option now triggers the Reset, setting the start time (and answer time
3579 if applicable) to the current time. Note that the 'a' option still sets
3580 the answer time to the current time if the channel was already answered.
3582 * The 's' option has been removed. A variable can be set on the original CDR
3583 if desired using the CDR function, and removed from a forked CDR using the
3586 * The 'T' option has been removed. The concept of DONT_TOUCH and LOCKED no
3587 longer applies in the CDR engine.
3589 * The 'v' option now prevents the copy of the variables from the original CDR
3590 to the forked CDR. Previously the variables were always copied but were
3591 removed from the original. This was changed as removing variables from a CDR
3592 can have unintended side effects - this option allows the user to prevent
3593 propagation of variables from the original to the forked without modifying
3598 * Added the 'n' option to MeetMe to prevent application of the DENOISE
3599 function to a channel joining a conference. Some channel drivers that vary
3600 the number of audio samples in a voice frame will experience significant
3601 quality problems if a denoiser is attached to the channel; this option gives
3602 them the ability to remove the denoiser without having to unload func_speex.
3606 * The 'b' option now includes conferences as well as sounds played to the
3609 * The AUDIOHOOK_INHERIT function is no longer needed to keep a MixMonitor
3610 running during a transfer. If a MixMonitor is started on a channel,
3611 the MixMonitor will continue to record the audio passing through the
3612 channel even in the presence of transfers.
3616 * The NoCDR application is deprecated. Please use the CDR_PROP function to
3619 * While the NoCDR application will prevent CDRs for a channel from being
3620 propagated to registered CDR backends, it will not prevent that data from
3621 being collected. Hence, a subsequent call to ResetCDR or the CDR_PROP
3622 function that enables CDRs on a channel will restore those records that have
3623 not yet been finalized.
3627 * The app_parkandannounce module has been removed. The application
3628 ParkAndAnnounce is now provided by the res_parking module. See the
3629 res_parking changes for more information.
3633 * Added queue available hint. The hint can be added to the dialplan using the
3634 following syntax: exten,hint,Queue:{queue_name}_avail
3635 For example, if the name of the queue is 'markq':
3636 exten => 8501,hint,Queue:markq_avail
3637 This will report 'InUse' if there are no logged in agents or no free agents.
3638 It will report 'Idle' when an agent is free.
3640 * Queues now support a hint for member paused state. The hint uses the form
3641 'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name}
3642 are the name of the queue and the name of the member to subscribe to,
3643 respectively. For example: exten => 8501,hint,Queue:sales_pause_mark.
3644 Members will show as In Use when paused.
3646 * The configuration options eventwhencalled and eventmemberstatus have been
3647 removed. As a result, the AMI events QueueMemberStatus, AgentCalled,
3648 AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be
3649 sent. The "Variable" fields will also no longer exist on the Agent* events.
3650 These events can be filtered out from a connected AMI client using the
3651 eventfilter setting in manager.conf.
3653 * The queue log now differentiates between blind and attended transfers. A
3654 blind transfer will result in a BLINDTRANSFER message with the destination
3655 context and extension. An attended transfer will result in an
3656 ATTENDEDTRANSFER message. This message will indicate the method by which
3657 the attended transfer was completed: "BRIDGE" for a bridge merge, "APP"
3658 for running an application on a bridge or channel, or "LINK" for linking
3659 two bridges together with local channels. The queue log will also now detect
3660 externally initiated blind and attended transfers and record the transfer
3663 * When performing queue pause/unpause on an interface without specifying an
3664 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
3665 least one member of any queue exists for that interface.
3667 * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
3668 for realtime queue log entries.
3672 * The 'e' option has been deprecated. Use the CDR_PROP function to re-enable
3673 CDRs when they were previously disabled on a channel.
3675 * The 'w' and 'a' options have been removed. Dispatching CDRs to registered
3676 backends occurs on an as-needed basis in order to preserve linkedid
3677 propagation and other needed behavior.
3681 * A new application, this is similar to SayAlpha except that it supports
3682 case sensitive playback of the specified characters. For example,
3683 SayAlphaCase(u,aBc) will result in 'a uppercase b c'.
3687 * This application is deprecated in favor of CHANNEL(amaflags).
3691 * The SendDTMF application will now accept 'W' as valid input. This will cause
3692 the application to delay one second while streaming DTMF.
3696 * A new application in Asterisk 12, this hands control of the channel calling
3697 the application over to an external system. Currently, external systems
3698 manipulate channels in Stasis through the Asterisk RESTful Interface (ARI).
3702 * UserEvent will now handle duplicate keys by overwriting the previous value
3703 assigned to the key.
3705 * In addition to AMI, UserEvent invocations will now be distributed to any
3706 interested Stasis applications.
3710 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
3711 system as mailbox@context. The rest of the system cannot add @default
3712 to mailbox identifiers for app_voicemail that do not specify a context
3713 any longer. It is a mailbox identifier format that should only be
3714 interpreted by app_voicemail.
3716 * The voicemail.conf configuration file now has an 'alias' configuration
3717 parameter for use with the Directory application. The voicemail realtime
3718 database table schema has also been updated with an 'alias' column.
3723 * Pass through support has been added for both VP8 and Opus.
3725 * Added format attribute negotiation for the Opus codec. Format attribute
3726 negotiation is provided by the res_format_attr_opus module.
3731 * Masquerades as an operation inside Asterisk have been effectively hidden
3732 by the migration to the Bridging API. As such, many 'quirks' of Asterisk
3733 no longer occur. This includes renaming of channels, "<ZOMBIE>" channels,
3734 dropping of frame/audio hooks, and other internal implementation details
3735 that users had to deal with. This fundamental change has large implications
3736 throughout the changes documented for this version. For more information
3737 about the new core architecture of Asterisk, please see the Asterisk wiki.
3739 * Multiple parties in a bridge may now be transferred. If a participant in a
3740 multi-party bridge initiates a blind transfer, a Local channel will be used
3741 to execute the dialplan location that the transferer sent the parties to. If
3742 a participant in a multi-party bridge initiates an attended transfer,
3743 several options are possible. If the attended transfer results in a transfer
3744 to an application, a Local channel is used. If the attended transfer results
3745 in a transfer to another channel, the resulting channels will be merged into
3748 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel
3749 driver specific. If the channel variable is set on the transferrer channel,
3750 the sound will be played to the target of an attended transfer.
3752 * The channel variable BRIDGEPEER becomes a comma separated list of peers in
3753 a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers
3754 listed. Any more peers in the bridge will not be included in the list.
3755 BRIDGEPEER is not valid in holding bridges like parking since those channels
3756 do not talk to each other even though they are in a bridge.
3758 * The channel variable BRIDGEPVTCALLID is only valid for two party bridges
3759 and will contain a value if the BRIDGEPEER's channel driver supports it.
3761 * A channel variable ATTENDEDTRANSFER is now set which indicates which channel
3762 was responsible for an attended transfer in a similar fashion to
3765 * Modules using the Configuration Framework or Sorcery must have XML
3766 configuration documentation. This configuration documentation is included
3767 with the rest of Asterisk's XML documentation, and is accessible via CLI
3768 commands. See the CLI changes for more information.
3770 AMI (Asterisk Manager Interface)
3772 * Major changes were made to both the syntax as well as the semantics of the
3773 AMI protocol. In particular, AMI events have been substantially improved
3774 in this version of Asterisk. For more information, please see the AMI
3775 specification at https://wiki.asterisk.org/wiki/x/dAFRAQ
3777 * AMI events that reference a particular channel or bridge will now always
3778 contain a standard set of fields. When multiple channels or bridges are
3779 referenced in an event, fields for at least some subset of the channels
3780 and bridges in the event will be prefixed with a descriptive name to avoid
3781 name collisions. See the AMI event documentation on the Asterisk wiki for
3784 * The CLI command 'manager show commands' no longer truncates command names
3785 longer than 15 characters and no longer shows authorization requirement
3786 for commands. 'manager show command' now displays the privileges needed
3787 for using a given manager command instead.
3789 * The SIPshowpeer action will now include a 'SubscribeContext' field for a
3790 peer in its response if the peer has a subscribe context set.
3792 * The SIPqualifypeer action now acknowledges the request once it has
3793 established that the request is against a known peer. It also issues a new
3794 event, 'SIPQualifyPeerDone', once the qualify action has been completed.
3796 * The PlayDTMF action now supports an optional 'Duration' parameter. This
3797 specifies the duration of the digit to be played, in milliseconds.
3799 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
3800 updates when changes occur instead of requiring the use of pollmailboxes.
3802 * Added a new action 'ControlPlayback'. The ControlPlayback action allows an
3803 AMI client to manipulate audio currently being played back on a channel. The
3804 supported operations depend on the application being used to send audio to
3805 the channel. When the audio playback was initiated using the ControlPlayback
3806 application or CONTROL STREAM FILE AGI command, the audio can be paused,
3807 stopped, restarted, reversed, or skipped forward. When initiated by other
3808 mechanisms (such as the Playback application), the audio can be stopped,
3809 reversed, or skipped forward.
3811 * Channel related events now contain a snapshot of channel state, adding new
3812 fields to many of these events.
3814 * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
3815 in a future release. Please use the common 'Exten' field instead.
3817 * The AMI event 'UserEvent' from app_userevent now contains the channel state
3818 fields. The channel state fields will come before the body fields.
3820 * The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
3821 'UnParkedCall' have changed significantly in the new res_parking module.
3823 The 'Channel' and 'From' headers are gone. For the channel that was parked
3824 or is coming out of parking, a 'Parkee' channel snapshot is issued and it
3825 has a number of fields associated with it. The old 'Channel' header relayed
3826 the same data as the new 'ParkeeChannel' header.
3828 The 'From' field was ambiguous and changed meaning depending on the event.
3829 for most of these, it was the name of the channel that parked the call
3830 (the 'Parker'). There is no longer a header that provides this channel name,
3831 however the 'ParkerDialString' will contain a dialstring to redial the
3832 device that parked the call.
3834 On UnParkedCall events, the 'From' header would instead represent the
3835 channel responsible for retrieving the parkee. It receives a channel
3836 snapshot labeled 'Retriever'. The 'from' field is is replaced with
3839 Lastly, the 'Exten' field has been replaced with 'ParkingSpace'.
3841 * The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar
3842 fashion has changed the field names 'StartExten' and 'StopExten' to
3843 'StartSpace' and 'StopSpace' respectively.
3845 * The deprecated use of | (pipe) as a separator in the channelvars setting in
3846 manager.conf has been removed.
3848 * Channel Variables conveyed with a channel no longer contain the name of the
3849 channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now
3850 ChanVariable: bar=baz. When multiple channels are present in a single AMI
3851 event, the various ChanVariable fields will contain a suffix that specifies
3852 which channel they correspond to.
3854 * The NewPeerAccount AMI event is no longer raised. The NewAccountCode AMI
3855 event always conveys the AMI event for a particular channel.
3857 * All 'Reload' events have been consolidated into a single event type. This
3858 event will always contain a Module field specifying the name of the module
3859 and a Status field denoting the result of the reload. All modules now issue
3860 this event when being reloaded.
3862 * The 'ModuleLoadReport' event has been removed. Most AMI connections would
3863 fail to receive this event due to being connected after modules have loaded.
3864 AMI connections that want to know when Asterisk is ready should listen for
3865 the 'FullyBooted' event.
3867 * app_fax now sends the same send fax/receive fax events as res_fax. The
3868 'FaxSent' event is now the 'SendFAX' event, and the 'FaxReceived' event is
3869 now the 'ReceiveFAX' event.
3871 * The 'MusicOnHold' event is now two events: 'MusicOnHoldStart' and
3872 'MusicOnHoldStop'. The sub type field has been removed.
3874 * The 'JabberEvent' event has been removed. It is not AMI's purpose to be a
3875 carrier for another protocol.
3877 * The Bridge Manager action's 'Playtone' header now accepts more fine-grained
3878 options. 'Channel1' and 'Channel2' may be specified in order to play a tone
3879 to the specific channel. 'Both' may be specified to play a tone to both
3880 channels. The old 'yes' option is still accepted as a way of playing the
3881 tone to Channel2 only.
3883 * The AMI 'Status' response event to the AMI Status action replaces the
3884 'BridgedChannel' and 'BridgedUniqueid' headers with the 'BridgeID' header to
3885 indicate what bridge the channel is currently in.
3887 * The AMI 'Hold' event has been moved out of individual channel drivers, into
3888 core, and is now two events: 'Hold' and 'Unhold'. The status field has been
3891 * The AMI events in app_queue have been made more consistent with each other.
3892 Events that reference channels (QueueCaller* and Agent*) will show
3893 information about each channel. The (infamous) 'Join' and 'Leave' AMI
3894 events have been changed to 'QueueCallerJoin' and 'QueueCallerLeave'.
3896 * The 'MCID' AMI event now publishes a channel snapshot when available and
3897 its non-channel-snapshot parameters now use either the "MCallerID" or
3898 'MConnectedID' prefixes with Subaddr*, Name*, and Num* suffixes instead
3899 of 'CallerID' and 'ConnectedID' to avoid confusion with similarly named
3900 parameters in the channel snapshot.
3902 * The AMI events 'Agentlogin' and 'Agentlogoff' have been renamed
3903 'AgentLogin' and 'AgentLogoff' respectively.
3905 * The 'Channel' key used in the 'AlarmClear', 'Alarm', and 'DNDState' has been
3906 renamed "DAHDIChannel" since it does not convey an Asterisk channel name.
3908 * 'ChannelUpdate' events have been removed.
3910 * All AMI events now contain a 'SystemName' field, if available.
3912 * Local channel optimization is now conveyed in two events:
3913 'LocalOptimizationBegin' and 'LocalOptimizationEnd'. The Begin event is sent
3914 when the Local channel driver begins attempting to optimize itself out of
3915 the media path; the End event is sent after the channel halves have
3916 successfully optimized themselves out of the media path.
3918 * Local channel information in events is now prefixed with 'LocalOne' and
3919 'LocalTwo'. This replaces the suffix of '1' and '2' for the two halves of
3920 the Local channel. This affects the 'LocalBridge', 'LocalOptimizationBegin',
3921 and 'LocalOptimizationEnd' events.
3923 * The option 'allowmultiplelogin' can now be set or overriden in a particular
3924 account. When set in the general context, it will act as the default
3925 setting for defined accounts.
3927 * The 'BridgeAction' event was removed. It technically added no value, as the
3928 Bridge Action already receives confirmation of the bridge through a
3929 successful completion Event.
3931 * The 'BridgeExec' events were removed. These events duplicated the events that
3932 occur in the Briding API, and are conveyed now through BridgeCreate,
3933 BridgeEnter, and BridgeLeave events.
3935 * The 'RTCPSent'/'RTCPReceived' events have been significantly modified from
3936 previous versions. They now report all SR/RR packets sent/received, and
3937 have been restructured to better reflect the data sent in a SR/RR. In
3938 particular, the event structure now supports multiple report blocks.
3940 * Added 'BlindTransfer' and 'AttendedTransfer' events. These events are
3941 raised when a blind transfer/attended transfer completes successfully.
3942 They contain information about the transfer that just completed, including
3943 the location of the transfered channel.
3945 * Added a 'security' class to AMI which outputs the required fields for
3946 security messages similar to the log messages from res_security_log
3948 * The AMI event 'ExtensionStatus' now contains a 'StatusText' field
3949 that describes the status value in a human readable string.
3951 CDR (Call Detail Records)
3953 * Significant changes have been made to the behavior of CDRs. The CDR engine
3954 was effectively rewritten and built on the Stasis message bus. For a full
3955 definition of CDR behavior in Asterisk 12, please read the specification
3956 on the Asterisk wiki (wiki.asterisk.org).
3958 * CDRs will now be created between all participants in a bridge. For each
3959 pair of channels in a bridge, a CDR is created to represent the path of
3960 communication between those two endpoints. This lets an end user choose who
3961 to bill for what during bridge operations with multiple parties.
3963 * The duration, billsec, start, answer, and end times now reflect the times
3964 associated with the current CDR for the channel, as opposed to a cumulative
3965 measurement of all CDRs for that channel.
3967 * When a CDR is dispatched, user defined CDR variables from both parties are
3968 included in the resulting CDR. If both parties have the same variable, only
3969 the Party A value is provided.
3971 * Added a new option to cdr.conf, 'debug'. When enabled, significantly more
3972 information regarding the CDR engine is logged as verbose messages. This
3973 option should only be used if the behavior of the CDR engine needs to be
3976 * Added CLI command 'cdr set debug {on|off}'. This toggles the 'debug' setting
3977 normally configured in cdr.conf.
3979 * Added CLI command 'cdr show active {channel}'. When {channel} is not
3980 specified, this command provides a summary of the channels with CDR
3981 information and their statistics. When {channel} is specified, it shows
3982 detailed information about all records associated with {channel}.
3984 CEL (Channel Event Logging)
3986 * CEL has undergone significant rework in Asterisk 12, and is now built on the
3987 Stasis message bus. Please see the specification for CEL on the Asterisk
3988 wiki at https://wiki.asterisk.org/wiki/x/4ICLAQ for more detailed
3991 * The 'extra' field of all CEL events that use it now consists of a JSON blob
3992 with key/value pairs which are defined in the Asterisk 12 CEL documentation.
3994 * BLINDTRANSFER events now report the transferee bridge unique
3995 identifier, extension, and context in a JSON blob as the extra string
3996 instead of the transferee channel name as the peer.
3998 * ATTENDEDTRANSFER events now report the peer as NULL and additional
3999 information in the 'extra' string as a JSON blob. For transfers that occur
4000 between two bridged channels, the 'extra' JSON blob contains the primary
4001 bridge unique identifier, the secondary channel name, and the secondary
4002 bridge unique identifier. For transfers that occur between a bridged channel
4003 and a channel running an app, the 'extra' JSON blob contains the primary
4004 bridge unique identifier, the secondary channel name, and the app name.
4006 * LOCAL_OPTIMIZE events have been added to convey local channel
4007 optimizations with the record occurring for the semi-one channel and
4008 the semi-two channel name in the peer field.
4010 * BRIDGE_START, BRIDGE_END, BRIDGE_UPDATE, 3WAY_START, 3WAY_END, CONF_ENTER,
4011 CONF_EXIT, CONF_START, and CONF_END events have all been removed. These
4012 events have been replaced by BRIDGE_ENTER/BRIDGE_EXIT. The BRIDGE_ENTER
4013 and BRIDGE_EXIT events are raised when a channel enters/exits any bridge,
4014 regardless of whether or not that bridge happens to contain multiple
4019 * When compiled with '--enable-dev-mode', the astobj2 library will now add
4020 several CLI commands that allow for inspection of ao2 containers that
4021 register themselves with astobj2. The CLI commands are 'astobj2 container
4022 dump', 'astobj2 container stats', and 'astobj2 container check'.
4024 * Added specific CLI commands for bridge inspection. This includes 'bridge
4025 show all', which lists all bridges in the system, and 'bridge show {id}',
4026 which provides specific information about a bridge.
4028 * Added CLI command 'bridge destroy'. This will destroy the specified bridge,
4029 ejecting the channels currently in the bridge. If the channels cannot
4030 continue in the dialplan or application that put them in the bridge, they
4033 * Added command 'bridge kick'. This will eject a single channel from a bridge.
4035 * Added commands to inspect and manipulate the registered bridge technologies.
4036 This include 'bridge technology show', which lists the registered bridge
4037 technologies, as well as 'bridge technology {suspend|unsuspend} {tech}',
4038 which controls whether or not a registered bridge technology can be used
4039 during smart bridge operations. If a technology is suspended, it will not
4040 be used when a bridge technology is picked for channels; when unsuspended,
4041 it can be used again.
4043 * The command 'config show help {module} {type} {option}' will show
4044 configuration documentation for modules with XML configuration
4045 documentation. When {module}, {type}, and {option} are omitted, a listing
4046 of all modules with registered documentation is displayed. When {module}
4047 is specified, a listing of all configuration types for that module is
4048 displayed, along with their synopsis. When {module} and {type} are
4049 specified, a listing of all configuration options for that type are
4050 displayed along with their synopsis. When {module}, {type}, and {option}
4051 are specified, detailed information for that configuration option is
4054 * Added 'core show sounds' and 'core show sound' CLI commands. These display
4055 a listing of all installed media sounds available on the system and
4056 detailed information about a sound, respectively.
4058 * 'xmldoc dump' has been added. This CLI command will dump the XML
4059 documentation DOM as a string to the specified file. The Asterisk core
4060 will populate certain XML elements pulled from the source files with
4061 additional run-time information; this command lets a user produce the
4062 XML documentation with all information.
4066 * Parking has been pulled from core and placed into a separate module called
4067 res_parking. See Parking changes below for more details. Configuration for
4068 parking should now be performed in res_parking.conf. Configuration for
4069 parking in features.conf is now unsupported.
4071 * Core attended transfers now have several new options. While performing an
4072 attended transfer, the transferer now has the following options:
4073 - *1 - cancel the attended transfer (configurable via atxferabort)
4074 - *2 - complete the attended transfer, dropping out of the call
4075 (configurable via atxfercomplete)
4076 - *3 - complete the attended transfer, but stay in the call. This will turn
4077 the call into a multi-party bridge (configurable via atxferthreeway)
4078 - *4 - swap to the other party. Once an attended transfer has begun, this
4079 options may be used multiple times (configurable via atxferswap)
4081 * For DTMF blind and attended transfers, the channel variable TRANSFER_CONTEXT
4082 must be on the channel initiating the transfer to have any effect.
4084 * The BRIDGE_FEATURES channel variable would previously only set features for
4085 the calling party and would set this feature regardless of whether the
4086 feature was in caps or in lowercase. Use of a caps feature for a letter
4087 will now apply the feature to the calling party while use of a lowercase
4088 letter will apply that feature to the called party.
4090 * Add support for automixmon to the BRIDGE_FEATURES channel variable.
4092 * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is
4093 removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that
4094 activated the dynamic feature.
4096 * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set
4097 only on the channel executing the dynamic feature. Executing a dynamic
4098 feature on the bridge peer in a multi-party bridge will execute it on all
4099 peers of the activating channel.
4101 * You can now have the settings for a channel updated using the FEATURE()
4102 and FEATUREMAP() functions inherited to child channels by setting
4103 FEATURE(inherit)=yes.
4105 * automixmon now supports additional channel variables from automon including:
4106 TOUCH_MIXMONITOR_PREFIX, TOUCH_MIXMONITOR_MESSAGE_START,
4107 and TOUCH_MIXMONITOR_MESSAGE_STOP
4109 * A new general features.conf option 'recordingfailsound' has been added which
4110 allowssetting a failure sound for a user tries to invoke a recording feature
4111 such as automon or automixmon and it fails.
4113 * It is no longer necessary (or possible) to define the ATXFER_NULL_TECH in
4114 features.c for atxferdropcall=no to work properly. This option now just
4119 * Added log rotation strategy 'none'. If set, no log rotation strategy will
4120 be used. Given that this can cause the Asterisk log files to grow quickly,
4121 this option should only be used if an external mechanism for log management
4126 * Dynamic realtime tables for SIP Users can now include a 'path' field. This
4127 will store the path information for that peer when it registers. Realtime
4128 tables can also use the 'supportpath' field to enable Path header support.
4130 * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
4131 objectIdentifier. This maps to the supportpath option in sip.conf.
4135 * Sorcery is a new data abstraction and object persistence API in Asterisk. It
4136 provides modules a useful abstraction on top of the many storage mechanisms
4137 in Asterisk, including the Asterisk Database, static configuration files,
4138 static Realtime, and dynamic Realtime. It also provides a caching service.
4139 Users can configure a hierarchy of data storage layers for specific modules
4142 * All future modules which utilize Sorcery for object persistence must have a
4143 column named "id" within their schema when using the Sorcery realtime module.
4144 This column must be able to contain a string of up to 128 characters in length.
4146 Security Events Framework
4148 * Security Event timestamps now use ISO 8601 formatted date/time instead of
4149 the "seconds-microseconds" format that it was using previously.
4153 * The Stasis message bus is a publish/subscribe message bus internal to
4154 Asterisk. Many services in Asterisk are built on the Stasis message bus,
4155 including AMI, ARI, CDRs, and CEL. Parameters controlling the operation of
4156 Stasis can be configured in stasis.conf. Note that these parameters operate
4157 at a very low level in Asterisk, and generally will not require changes.
4161 * When a channel driver is configured to enable jiterbuffers, they are now
4162 applied unconditionally when a channel joins a bridge. If a jitterbuffer
4163 is already set for that channel when it enters, such as by the JITTERBUFFER
4164 function, then the existing jitterbuffer will be used and the one set by
4165 the channel driver will not be applied.
4169 * chan_agent has been removed and replaced with AgentLogin and AgentRequest
4170 dialplan applications provided by the app_agent_pool module. Agents are
4171 connected with callers using the new AgentRequest dialplan application.
4172 The Agents:<agent-id> device state is available to monitor the status of an
4173 agent. See agents.conf.sample for valid configuration options.
4175 * The updatecdr option has been removed. Altering the names of channels on a
4176 CDR is not supported - the name of the channel is the name of the channel,
4177 and pretending otherwise helps no one. The AGENTUPDATECDR channel variable
4178 has also been removed, for the same reason.
4180 * The endcall and enddtmf configuration options are removed. Use the
4181 dialplan function CHANNEL(dtmf_features) to set DTMF features on the agent
4182 channel before calling AgentLogin.
4186 * chan_bridge has been removed. Its functionality has been incorporated
4187 directly into the ConfBridge application itself.
4191 * Added the CLI command 'pri destroy span'. This will destroy the D-channel
4192 of the specified span and its B-channels. Note that this command should
4193 only be used if you understand the risks it entails.
4195 * The CLI command 'dahdi destroy channel' is now 'dahdi destroy channels'.
4196 A range of channels can be specified to be destroyed. Note that this command
4197 should only be used if you understand the risks it entails.
4199 * Added the CLI command 'dahdi create channels'. A range of channels can be
4200 specified to be created, or the keyword 'new' can be used to add channels
4203 * The script specified by the chan_dahdi.conf mwimonitornotify option now gets
4204 the exact configured mailbox name. For app_voicemail mailboxes this is
4207 * Added mwi_vm_boxes that also must be configured for ISDN MWI to be enabled.
4211 * IPv6 support has been added. We are now able to bind to and
4212 communicate using IPv6 addresses.
4216 * The /b option has been removed.
4218 * chan_local moved into the system core and is no longer a loadable module.
4222 * Added general support for busy detection.
4224 * Added ECAM command support for Sony Ericsson phones.
4228 * A new SIP channel driver for Asterisk, chan_pjsip is built on the PJSIP
4229 SIP stack. A collection of resource modules provides the bulk of the SIP
4230 functionality. For more information on the new SIP channel driver, see
4231 https://wiki.asterisk.org/wiki/x/JYGLAQ
4235 * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
4236 using the 'supportpath' setting, either on a global basis or on a peer basis.
4237 This setting enables Asterisk to route outgoing out-of-dialog requests via a
4238 set of proxies by using a pre-loaded route-set defined by the Path headers in
4239 the REGISTER request. See Realtime updates for more configuration information.
4241 * The SIP_CODEC family of variables may now specify more than one codec. Each
4242 codec must be separated by a comma. The first codec specified is the
4243 preferred codec for the offer. This allows a dialplan writer to specify both
4244 audio and video codecs, e.g., Set(SIP_CODEC=ulaw,h264)
4246 * The 'callevents' parameter has been removed. Hold AMI events are now raised
4247 in the core, and can be filtered out using the 'eventfilter' parameter
4250 * Added 'ignore_requested_pref'. When enabled, this will use the preferred
4251 codecs configured for a peer instead of the requested codec.
4253 * The option "register_retry_403" has been added to chan_sip to work around
4254 servers that are known to erroneously send 403 in response to valid
4255 REGISTER requests and allows Asterisk to continue attepmting to connect.
4259 * Added the 'immeddialkey' parameter. If set, when the user presses the
4260 configured key the already entered number will be immediately dialed. This
4261 is useful when the dialplan allows for variable length pattern matching.
4262 Valid options are '*' and '#'.
4264 * Added the 'callfwdtimeout' parameter. This configures the amount of time (in
4265 milliseconds) before a call forward is considered to not be answered.
4267 * The 'serviceurl' parameter allows Service URLs to be attached to line
4276 * The password option has been disabled, as the AgentLogin application no
4277 longer provides authentication.
4281 * Due to changes in the Asterisk core, this function is no longer needed to
4282 preserve a MixMonitor on a channel during transfer operations and dialplan
4283 execution. It is effectively obsolete.
4287 * The 'amaflags' and 'accountcode' attributes for the CDR function are
4288 deprecated. Use the CHANNEL function instead to access these attributes.
4290 * The 'l' option has been removed. When reading a CDR attribute, the most
4291 recent record is always used. When writing a CDR attribute, all non-finalized
4294 * The 'r' option has been removed, for the same reason as the 'l' option.
4296 * The 's' option has been removed, as LOCKED semantics no longer exist in the
4301 * A new function CDR_PROP has been added. This function lets you set properties
4302 on a channel's active CDRs. This function is write-only. Properties accept
4303 boolean values to set/clear them on the channel's CDRs. Valid properties
4305 - 'party_a' - make this channel the preferred Party A in any CDR between two
4306 channels. If two channels have this property set, the creation time of the
4307 channel is used to determine who is Party A. Note that dialed channels are
4308 never Party A in a CDR.
4309 - 'disable' - disable CDRs on this channel. This is analogous to the NoCDR
4310 application when set to True, and analogous to the 'e' option in ResetCDR
4315 * Added the argument 'dtmf_features'. This sets the DTMF features that will be
4316 enabled on a channel when it enters a bridge. Allowed values are 'T', 'K',
4317 'H', 'W', and 'X', and are analogous to the parameters passed to the Dial
4320 * Added the argument 'after_bridge_goto'. This can be set to a parseable Goto
4321 string, i.e., [[context],extension],priority. If set on a channel, if a
4322 channel leaves a bridge but is not hung up it will resume dialplan execution
4327 * JITTERBUFFER now accepts an argument of 'disabled' which can be used
4328 to remove jitterbuffers previously set on a channel with JITTERBUFFER.
4329 The value of this setting is ignored when disabled is used for the argument.
4333 * A new function provided by chan_pjsip, this function can be used in
4334 conjunction with the Dial application to construct a dial string that will
4335 dial all contacts on an Address of Record associated with a chan_pjsip
4340 * Provided by chan_pjsip, this function sets the codecs to be offerred on the
4341 outbound channel prior to dialing.
4345 * Redirecting reasons can now be set to arbitrary strings. This means
4346 that the REDIRECTING dialplan function can be used to set the redirecting
4347 reason to any string. It also allows for custom strings to be read as the
4348 redirecting reason from SIP Diversion headers.
4352 * The SPEECH_ENGINE function now supports read operations. When read from, it
4353 will return the current value of the requested attribute.
4357 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
4358 system as mailbox@context. The rest of the system cannot add @default
4359 to mailbox identifiers for app_voicemail that do not specify a context
4360 any longer. It is a mailbox identifier format that should only be
4361 interpreted by app_voicemail.
4367 res_agi (Asterisk Gateway Interface)
4369 * The manager event AGIExec has been split into AGIExecStart and AGIExecEnd.
4371 * The manager event AsyncAGI has been split into AsyncAGIStart, AsyncAGIExec,
4374 * The CONTROL STREAM FILE command now accepts an offsetms parameter. This
4375 will start the playback of the audio at the position specified. It will
4376 also return the final position of the file in 'endpos'.
4378 * The CONTROL STREAM FILE command will now populate the CPLAYBACKSTATUS
4379 channel variable if the user stopped the file playback or if a remote
4380 entity stopped the playback. If neither stopped the playback, it will
4381 indicate the overall success/failure of the playback. If stopped early,
4382 the final offset of the file will be set in the CPLAYBACKOFFSET channel
4385 * The SAY ALPHA command now accepts an additional parameter to control
4386 whether it specifies the case of uppercase, lowercase, or all letters to
4387 provide functionality similar to SayAlphaCase.
4389 res_ari (Asterisk RESTful Interface) (and others)
4391 * The Asterisk RESTful Interface (ARI) provides a mechanism to expose and
4392 control telephony primitives in Asterisk by remote client. This includes
4393 channels, bridges, endpoints, media, and other fundamental concepts. Users
4394 of ARI can develop their own communications applications, controlling
4395 multiple channels using an HTTP RESTful interface and receiving JSON events
4396 about the objects via a WebSocket connection. ARI can be configured in
4397 Asterisk via ari.conf. For more information on ARI, see
4398 https://wiki.asterisk.org/wiki/x/0YCLAQ
4402 * Parking has been extracted from the Asterisk core as a loadable module,
4403 res_parking. Configuration for parking is now provided by res_parking.conf.
4404 Configuration through features.conf is no longer supported.
4406 * res_parking uses the configuration framework. If an invalid configuration is
4407 supplied, res_parking will fail to load or fail to reload. Previously,
4408 invalid configurations would generally be accepted, with certain errors
4409 resulting in individually disabled parking lots.
4411 * Parked calls are now placed in bridges. While this is largely an
4412 architectural change, it does have implications on how channels in a parking
4413 lot are viewed. For example, commands that display channels in bridges will
4414 now also display the channels in a parking lot.
4416 * The order of arguments for the new parking applications have been modified.
4417 Timeout and return context/exten/priority are now implemented as options,
4418 while the name of the parking lot is now the first parameter. See the
4419 application documentation for Park, ParkedCall, and ParkAndAnnounce for more
4420 in-depth information as well as syntax.
4422 * Extensions are by default no longer automatically created in the dialplan to
4423 park calls or pickup parked calls. Generation of dialplan extensions can be
4424 enabled using the 'parkext' configuration option.
4426 * ADSI functionality for parking is no longer supported. The 'adsipark'
4427 configuration option has been removed as a result.
4429 * The PARKINGSLOT channel variable has been deprecated in favor of
4430 PARKING_SPACE to match the naming scheme of the new system.
4432 * PARKING_SPACE and PARKEDLOT channel variables will now be set for a parked
4433 channel even when the configuration option 'comebactoorigin' is enabled.
4435 * A new CLI command 'parking show' has been added. This allows a user to
4436 inspect the parking lots that are currently in use.
4437 'parking show <parkinglot>' will also show the parked calls in a specific
4440 * The CLI command 'parkedcalls' is now deprecated in favor of
4441 'parking show <parkinglot>'.
4443 * The AMI command 'ParkedCalls' will now accept a 'ParkingLot' argument which
4444 can be used to get a list of parked calls for a specific parking lot.
4446 * The AMI command 'Park' field 'Channel2' has been deprecated and replaced
4447 with 'TimeoutChannel'. If both 'Channel2' and 'TimeoutChannel' are
4448 specified, 'TimeoutChannel' will be used. The field 'TimeoutChannel' is no
4449 longer a required argument.
4451 * The ParkAndAnnounce application is now provided through res_parking instead
4452 of through the separate app_parkandannounce module.
4454 * ParkAndAnnounce will no longer go to the next position in dialplan on timeout
4455 by default. Instead, it will follow the timeout rules of the parking lot. The
4456 old behavior can be reproduced by using the 'c' option.
4458 * Dynamic parking lots will now fail to be created under the following
4460 - if the parking lot specified by PARKINGDYNAMIC does not exist
4461 - if they require exclusive park and parkedcall extensions which overlap
4462 with existing parking lots.
4464 * Dynamic parking lots will be cleared on reload for dynamic parking lots that
4465 currently contain no calls. Dynamic parking lots containing parked calls
4466 will persist through the reloads without alteration.
4468 * If 'parkext_exclusive' is set for a parking lot and that extension is
4469 already in use when that parking lot tries to register it, this is now
4470 considered a parking system configuration error. Configurations which do
4471 this will be rejected.
4473 * Added channel variable PARKER_FLAT. This contains the name of the extension
4474 that would be used if 'comebacktoorigin' is enabled. This can be useful when
4475 comebacktoorigin is disabled, but the dialplan or an external control
4476 mechanism wants to use the extension in the park-dial context that was
4477 generated to re-dial the parker on timeout.
4479 res_pjsip (and many others)
4481 * A large number of resource modules make up the SIP stack based on pjsip.
4482 The chan_pjsip channel driver users these resource modules to provide
4483 various SIP functionality in Asterisk. The majority of configuration for
4484 these modules is performed in pjsip.conf. Other modules may use their
4485 own configuration files.
4487 * Added 'set_var' option for an endpoint. For each variable specified that
4488 variable gets set upon creation of a channel involving the endpoint.
4492 * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
4493 them, an Asterisk-specific version of PJSIP needs to be installed.
4494 Tarballs are available from https://github.com/asterisk/pjproject/tags/.
4496 res_statsd/res_chan_stats
4498 * A new resource module, res_statsd, has been added, which acts as a statsd
4499 client. This module allows Asterisk to publish statistics to a statsd
4500 server. In conjunction with res_chan_stats, it will publish statistics about
4501 channels to the statsd server. It can be configured via res_statsd.conf.
4505 * Device state for XMPP buddies is now available using the following format:
4506 XMPP/<client name>/<buddy address>
4507 If any resource is available the device state is considered to be not in use.
4508 If no resources exist or all are unavailable the device state is considered
4515 Realtime/Database Scripts
4517 * Asterisk previously included example db schemas in the contrib/realtime/
4518 directory of the source tree. This has been replaced by a set of database
4519 migrations using the Alembic framework. This allows you to use alembic to
4520 initialize the database for you. It will also serve as a database migration
4521 tool when upgrading Asterisk in the future.
4523 See contrib/ast-db-manage/README.md for more details.
4527 * A new script has been added in the contrib/scripts/sip_to_res_pjsip folder.
4528 This python script will convert an existing sip.conf file to a
4529 pjsip.conf file, for use with the chan_pjsip channel driver. This script
4530 is meant to be an aid in converting an existing chan_sip configuration to
4531 a chan_pjsip configuration, but it is expected that configuration beyond
4532 what the script provides will be needed.
4534 ------------------------------------------------------------------------------
4535 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
4536 ------------------------------------------------------------------------------
4540 * The Asterisk build system will now build and install a shared library
4541 (libasteriskssl.so) used to wrap various initialization and shutdown functions
4542 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
4543 that Asterisk can ensure that these functions do *not* get called by any
4544 modules that are loaded into Asterisk, since they should only be called once
4545 in any single process. If desired, this feature can be disabled by supplying
4546 the "--disable-asteriskssl" option to the configure script.
4548 * A new make target, 'full', has been added to the Makefile. This performs
4549 the same compilation actions as make all, but will also scan the entirety of
4550 each source file for documentation. This option is needed to generate AMI
4551 event documentation. Note that your system must have Python in order for
4552 this make target to succeed.
4554 * The optimization portion of the build system has been reworked to avoid
4555 broken builds on certain architectures. All architecture-specific
4556 optimization has been removed in favor of using -march=native to allow gcc
4557 to detect the environment in which it is running when possible. This can
4558 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
4560 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
4561 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
4563 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
4564 previously parsed the header file to obtain the version of Asterisk, you
4565 will now have to go through Asterisk to get the version information.
4573 * Added 'F()' option. Similar to the dial option, this can be supplied with
4574 arguments indicating where the callee should go after the caller is hung up,
4575 or without options specified, the priority after the Queue will be used.
4580 * Added menu action admin_toggle_mute_participants. This will mute / unmute
4581 all non-admin participants on a conference. The confbridge configuration
4582 file also allows for the default sounds played to all conference users when
4583 this occurs to be overriden using sound_participants_unmuted and
4584 sound_participants_muted.
4586 * Added menu action participant_count. This will playback the number of
4587 current participants in a conference.
4589 * Added announcement configuration option to user profile. If set the sound
4590 file will be played to the user, and only the user, upon joining the
4593 * Added record_file_append option that defaults to "yes", but if set to no
4594 will create a new file between each start/stop recording.
4599 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
4600 channels respectively before the callee channels are called.
4605 * Added support for IPv6.
4607 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
4608 external process will cause the current playlist to be cleared, including
4609 stopping any audio file that is currently playing. This is useful when you
4610 want to interrupt audio playback only when specific DTMF is entered by the
4616 * A new option, 'I' has been added to app_followme. By setting this option,
4617 Asterisk will not update the caller with connected line changes when they
4618 occur. This is similar to app_dial and app_queue.
4620 * The 'N' option is now ignored if the call is already answered.
4622 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
4623 and caller channels respectively before the callee channels are called.
4625 * The winning FollowMe outgoing call is now put on hold if the caller put it on
4631 * MixMonitor hooks now have IDs associated with them which can be used to
4632 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
4633 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
4634 now accepts that ID as an argument.
4636 * Added 'm' option, which stores a copy of the recording as a voicemail in the
4637 indicated mailboxes.
4642 * The connect action in app_mysql now allows you to specify a port number to
4643 connect to. This is useful if you run a MySQL server on a non-standard
4649 * Increased the default number of allowed destinations from 5 to 12.
4654 * The app_page application now no longer depends on DAHDI or app_meetme. It
4655 has been re-architected to use app_confbridge internally.
4660 * Added queue options autopausebusy and autopauseunavail for automatically
4661 pausing a queue member when their device reports busy or congestion.
4663 * The 'ignorebusy' option for queue members has been deprecated in favor of
4664 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
4665 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
4666 per interface basis. Individual ringinuse values can now be set in
4667 queues.conf via an argument to member definitions. Lastly, the queue
4668 'ringinuse' setting now only determines defaults for the per member
4669 'ringinuse' setting and does not override per member settings like it does
4670 in earlier versions.
4672 * Added 'F()' option. Similar to the dial option, this can be supplied with
4673 arguments indicating where the callee should go after the caller is hung up,
4674 or without options specified, the priority after the Queue will be used.
4676 * Added new option log_member_name_as_agent, which will cause the membername to
4677 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
4678 state_interface has been set.
4680 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
4682 * App_queue will now play periodic announcements for the caller that
4683 holds the first position in the queue while waiting for answer.
4687 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
4688 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
4689 changed arguments to SayUnixTime so that every option is truly optional even
4690 when using multiple options (so that j option could be used without having to
4691 manually specify timezone and format) There are other benefits, e.g., format
4692 can now be used without specifying time zone as well.
4697 * Addition of the VM_INFO function - see Function changes.
4699 * The imapserver, imapport, and imapflags configuration options can now be
4700 overriden on a user by user basis.
4702 * When voicemail plays a message's envelope with saycid set to yes, when
4703 reaching the caller id field it will play a recording of a file with the same
4704 base name as the sender's callerid if there is a similarly named file in
4705 <astspooldir>/recordings/callerids/
4707 * Voicemails now contains a unique message identifier "msg_id", which is stored
4708 in the message envelope with the sound files. IMAP backends will now store
4709 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
4710 backends will store the message identifier in a "msg_id" column. See
4711 UPGRADE.txt for more information.
4713 * Added VoiceMailPlayMsg application. This application will play a single
4714 voicemail message from a mailbox. The result of the application, SUCCESS or
4715 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
4720 * Hangup handlers can be attached to channels using the CHANNEL() function.
4721 Hangup handlers will run when the channel is hung up similar to the h
4722 extension. The hangup_handler_push option will push a GoSub compatible
4723 location in the dialplan onto the channel's hangup handler stack. The
4724 hangup_handler_pop option will remove the last added location, and optionally
4725 replace it with a new GoSub compatible location. The hangup_handler_wipe
4726 option will remove all locations on the stack, and optionally add a new
4729 * The expression parser now recognizes the ABS() absolute value function,
4730 which will convert negative floating point values to positive values.
4732 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
4733 control of faxdetect.
4735 * Addition of the VM_INFO function that can be used to retrieve voicemail
4736 user information, such as the email address and full name.
4737 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
4740 * The REDIRECTING function now supports the redirecting original party id
4743 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
4744 lets you set some of the configuration options from the [general] section
4745 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
4746 the key sequence used to activate built-in features, such as blindxfer,
4747 and automon. See the built-in documentation for details.
4749 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
4750 instead of simply the uri. This is the format that MessageSend() can use
4751 in the from parameter for outgoing SIP messages.
4753 * Added the PRESENCE_STATE function. This allows retrieving presence state
4754 information from any presence state provider. It also allows setting
4755 presence state information from a CustomPresence presence state provider.
4756 See AMI/CLI changes for related commands.
4758 * Added the AMI_CLIENT function to make manager account attributes available
4759 to the dialplan. It currently supports returning the current number of
4760 active sessions for a given account.
4762 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
4763 and the REDIRECTING functions.
4771 * Added a manager event "LocalBridge" for local channel call bridges between
4772 the two pseudo-channels created.
4777 * Added dialtone_detect option for analog ports to disconnect incoming
4778 calls when dialtone is detected.
4780 * Added option colp_send to send ISDN connected line information. Allowed
4781 settings are block, to not send any connected line information; connect, to
4782 send connected line information on initial connect; and update, to send
4783 information on any update during a call. Default is update.
4785 * Add options namedcallgroup and namedpickupgroup to support installations
4786 where a higher number of groups (>64) is required.
4788 * Added support to use private party ID information with PRI calls.
4793 * A new channel driver named chan_motif has been added which provides support for
4794 Google Talk and Jingle in a single channel driver. This new channel driver includes
4795 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
4796 hold, unhold, and ringing notification. It is also compliant with the current Jingle
4797 specification, current Google Jingle specification, and the original Google Talk
4803 * Added NAT support for RTP. Setting in config is 'nat', which can be set
4804 globally and overriden on a peer by peer basis.
4806 * Direct media functionality has been added. Options in config are:
4807 directmedia (directrtp) and directrtpsetup (earlydirect)
4809 * ChannelUpdate events now contain a CallRef header.
4814 * Asterisk will no longer substitute CID number for CID name in the display
4815 name field if CID number exists without a CID name. This change improves
4816 compatibility with certain device features such as Avaya IP500's directory
4819 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
4820 created using that setting to not be removed during SIP reload.
4822 * Added settings recordonfeature and recordofffeature. When receiving an INFO
4823 request with a "Record:" header, this will turn the requested feature on/off.
4824 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
4825 dynamic features must be enabled and configured properly on the requesting
4826 channel for this to function properly.
4828 * Add support to realtime for the 'callbackextension' option.
4830 * When multiple peers exist with the same address, but differing
4831 callbackextension options, incoming requests that are matched by address
4832 will be matched to the peer with the matching callbackextension if it is
4835 * Two new NAT options, auto_force_rport and auto_comedia, have been added
4836 which set the force_rport and comedia options automatically if Asterisk
4837 detects that an incoming SIP request crossed a NAT after being sent by
4838 the remote endpoint.
4840 * The default global nat setting in sip.conf has been changed from force_rport
4841 to auto_force_rport.
4843 * NAT settings are now a combinable list of options. The equivalent of the
4844 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
4846 * Adds an option send_diversion which can be disabled to prevent
4847 diversion headers from automatically being added to INVITE requests.
4849 * Add support for lightweight NAT keepalive. If enabled a blank packet will
4850 be sent to the remote host at a given interval to keep the NAT mapping open.
4851 This can be enabled using the keepalive configuration option.
4853 * Add option 'tonezone' to specify country code for indications. This option
4854 can be set both globally and overridden for specific peers.
4856 * The SIP Security Events Framework now supports IPv6.
4858 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
4859 between multiple user agents. When set, for directmedia reinvites,
4860 Asterisk will not send an immediate reinvite on an incoming call leg. This
4861 option is useful when peered with another SIP user agent that is known to
4862 send immediate direct media reinvites upon call establishment.
4864 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
4867 * Add options subminexpiry and submaxexpiry to set limits of subscription
4868 timer independently from registration timer settings. The setting of the
4869 registration timer limits still is done by options minexpiry, maxexpiry
4870 and defaultexpiry. For backwards compatibility the setting of minexpiry
4871 and maxexpiry also is used to configure the subscription timer limits if
4872 subminexpiry and submaxexpiry are not set in sip.conf.
4874 * Set registration timer limits to default values when reloading sip
4875 configuration and values are not set by configuration.
4877 * Add options namedcallgroup and namedpickupgroup to support installations
4878 where a higher number of groups (>64) is required.
4880 * When a MESSAGE request is received, the address the request was received from
4881 is now saved in the SIP_RECVADDR variable.
4883 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
4884 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
4885 the ANI2/OLI information is set on the channel, which can be retrieved using
4886 the CALLERID function.
4888 * Peers can now be configured to support negotiation of ICE candidates using
4889 the setting icesupport. See res_rtp_asterisk changes for more information.
4891 * Added support for format attribute negotiation. See the Codecs changes for
4894 * Extra headers specified with SIPAddHeader are sent with the REFER message
4895 when using Transfer application. See refer_addheaders in sip.conf.sample.
4897 * Added support to use private party ID information with calls.
4899 * Adds an option discard_remote_hold_retrieval that when set stops telling
4900 the peer to start music on hold.
4905 * Added skinny version 17 protocol support.
4909 --------------------
4910 * Added option 'dtmf_duration' allowing playback time of DTMF tones to be set
4912 * Modified option 'date_format' to allow options to display date in 31Jan and Jan31
4913 formats as options 0 and 1. The previous options 0 and 1 now map to options 2 and 3
4914 as per the UNISTIM protocol.
4916 * Fixed issues with dialtone not matching indications.conf and mute stopping rx
4917 as well as tx. Also fixed issue with call "Timer" displaying as French "Dur\E9e"
4919 * Added ability to use multiple lines for a single phone. This allows multiple
4920 calls to occur on a single phone, using callwaiting and switching between calls.
4922 * Added option 'sharpdial' allowing end dialing by pressing # key
4924 * Added option 'interdigit_timer' to control phone dial timeout
4926 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
4928 * Added global 'debug' option, that enables debug in channel driver
4930 * Added ability to translate on-screen menu in multiple languages. Tested on
4931 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
4932 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
4935 * In addition to English added French and Russian languages for on-screen menus
4937 * Reworked dialing number input: added dialing by timeout, immediate dial on
4938 on dialplan compare,
projects / asterisk/asterisk.git / blob