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 --- Functionality changes from Asterisk 18.0.0 to Asterisk 19.0.0 ------------
17 ------------------------------------------------------------------------------
21 * Hook flash events are now exposed as AMI events.
23 Add variable support to Originate
25 * The Originate application now allows
26 variables to be set on the new channel
31 * Added debug logging categories that allow a user to output debug information
32 based on a specified category. This lets the user limit, and filter debug
33 output to data relevant to a particular context, or topic. For instance the
34 following categories are now available for debug logging purposes:
36 dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet, stun, stun_packet
38 These debug categories can be enable/disable via an Asterisk CLI command:
40 core set debug category <category>[:<sublevel>] [category[:<sublevel] ...]
41 core set debug category off [<category> [<category>] ...]
43 If no sub-level is associated all debug statements for a given category are
44 output. If a sub-level is given then only those statements assigned a value
45 at or below the associated sub-level are output.
47 * The location where the media cache stores its temporary files
48 is no longer hardcoded to /tmp but can now be configured separately
49 via the astcachedir config variable in asterisk.conf.
51 The default location for astcachedir is now /var/cache/asterisk
52 instead of /tmp, please make sure to manually cleanup and/or
53 migrate the temporary files in /tmp after upgrading.
57 * The MessageSend dialplan application now takes an
58 optional third argument that can set the message's
59 "To" field on outgoing messages. It's an alternative
60 to using the MESSAGE(to) dialplan function.
62 To prevent confusion with the first argument, currently
63 named "to", it's been renamed to "destination".
64 Its function, creating the request URI, hasn't changed.
66 The online documentation has also been enhanced to
69 Despite the changes in this commit, there should be
70 no impact to current users of MessageSend.
72 New ConfKick application
74 * Adds a ConfKick() application, which allows
75 a specific channel, all users, or all non-admin
76 users to be kicked from a conference bridge.
78 New Reload application
80 * Adds an application to reload modules
82 PlaybackFinished has a new error state
84 * The PlaybackFinished event now has a new state "failed"
85 that is used when the sound file was not played due to an error.
86 Before the state on PlaybackFinished was always "done".
88 In case of multiple sound files to be played,
89 the PlaybackFinished is sent only once in the end of the list,
90 even in case of error.
92 WaitForCondition application
94 * This application provides a way to halt
95 dialplan execution until a provided
96 condition evaluates to true.
100 * app_confbridge now has the ability to force the estimated bitrate on an SFU
101 bridge. To use it, set a bridge profile's remb_behavior to "force" and
102 set remb_estimated_bitrate to a rate in bits per second. The
103 remb_estimated_bitrate parameter is ignored if remb_behavior is something
106 app_confbridge answer supervision control
108 * app_confbridge now provides a user option to prevent
109 answer supervision if the channel hasn't been
110 answered yet. To use it, set a user profile's
111 answer_channel option to no.
113 app_dial announcement option
115 * The A option for Dial now supports
116 playing audio to the caller as well
121 * app_mixmonitor now sends manager events MixMonitorStart, MixMonitorStop and
122 MixMonitorMute when the channel monitoring is started, stopped and muted (or
123 unmuted) respectively.
127 * The VoiceMail application can now be configured to send greetings and
128 instructions via early media and only answering the channel when it is
129 time for the caller to record their message. This behavior can be
130 activated by passing the new 'e' option to VoiceMail.
132 * You can now customize the "beep" tone or omit it entirely.
136 * You can now specify a default "auth" method in the
137 [general] section of iax.conf
141 * The PJSIP_SEND_SESSION_REFRESH dialplan function now issues a warning, and
142 returns unsuccessful if it's used on a channel prior to answering.
144 chan_pjsip, app_transfer
146 * Added TRANSFERSTATUSPROTOCOL variable. When transfer is performed,
147 transfers can pass a protocol specific error code.
148 Example, in SIP 3xx-6xx represent any SIP specific error received when
151 func_math: Three new dialplan functions
153 * Introduce three new functions, MIN, MAX, and ABS, which can be used to
154 obtain the minimum or maximum of up to two integers or absolute value.
158 * Introduce an ARGC variable for func_odbc functions, along with a minargs
159 per-function configuration option.
161 minargs enables enforcing of minimum count of arguments to pass to
162 func_odbc, so if you're unconditionally using ARG1 through ARG4 then
163 this should be set to 4. func_odbc will generate an error in this case,
169 and ODBC_FOO(a,b,c) in dialplan will now error out instead of using a
170 potentially leaked ARG4 from Gosub().
172 ARGC is needed if you're using optional argument, to verify whether or
173 not an argument has been passed, else it's possible to use a leaked ARGn
174 from Gosub (app_stack). So now you can safely do
175 ${IF($[${ARGC}>3]?${ARGV}:default value)} kind of thing.
177 func_volume now can be read
179 * The VOLUME function can now also be used
180 to read existing values previously set.
184 * Added a new log formatter called "plain" that always prints
185 file, function and line number if available (even for verbose
186 messages) and never prints color control characters. Most
187 suitable for file output but can be used for other channels
190 You use it in logger.conf like so:
191 debug => [plain]debug
192 console => [plain]error,warning,debug,notice,pjsip_history
193 messages => [plain]warning,error,verbose
195 * The dateformat option in logger.conf will now control the remote
196 console (asterisk -r -T) timestamp format. Previously, dateformat only
197 controlled the formatting of the timestamp going to log files and the
198 main console (asterisk -c) but only for non-verbose messages.
200 Internally, Asterisk does not send the logging timestamp with verbose
201 messages to console clients. It's up to the Asterisk remote consoles
202 to format verbose messages. Asterisk remote consoles previously did
203 not load dateformat from logger.conf.
205 Previously there was a non-configurable and hard-coded "%b %e %T"
206 dateformat that would be used no matter what on all verbose console
207 messages printed on remote consoles.
214 [2021-03-19 09:54:19.760-0400] Loading res_stasis_answer.so.
215 [Mar 19 09:55:43] -- Goto (dialExten,s,1)
217 Given the following example configuration in logger.conf, Asterisk log
218 files and the console, will log verbose messages using the given
219 timestamp. Now ensuring that all remote console messages are logged
220 with the same dateformat as other log streams.
227 console => notice,warning,error,verbose
228 full => notice,warning,error,debug,verbose
231 Now we have a globally-defined dateformat that will be used
232 consistently across the Asterisk main console, remote consoles, and
235 Now we have consistent logging:
238 [2021-03-19 09:54:19.760-0400] Loading res_stasis_answer.so.
239 [2021-03-19 09:55:43.920-0400] -- Goto (dialExten,s,1)
243 * PJSIP transports can now be partially reloaded safely. This allows the
244 local_net and external_* options to be updated without restarting Asterisk.
246 * PJSIP endpoints can now be configured to skip authentication when
247 handling OPTIONS requests by setting the allow_unauthenticated_options
248 configuration property to 'yes.'
250 * PJSIP support of registrations of endpoints in multidomain
251 scenarios, where the endpoint contains the domain info
254 res_pjsip_dialog_info_body_generator
256 * PJSIP now supports RFC 4235 Section 4.1.6 dialog-info+xml local and
257 remote elements by iterating through ringing channels and inserting
258 that info into NOTIFY packet sent to the endpoint.
262 * Implemented the new "to" parameter of the MessageSend()
263 dialplan application. This allows a user to specify
264 a complete SIP "To" header separate from the Request URI.
265 We now also accept a destination in the same format
266 as Dial()... PJSIP/number@endpoint
270 * By default Asterisk reports the PJSIP version in all
271 STUN packets it sends.
273 This behaviour may not be desired in a production
274 environment and can now be disabled by setting the
275 stun_software_attribute option to 'no' in rtp.conf.
279 * SRTP replay protection has been added to res_srtp and
280 a new configuration option "srtpreplayprotection" has
281 been added to the rtp.conf config file. For security
282 reasons, the default setting is "yes". Buggy clients
283 may not handle this correctly which could result in
284 no, or one way, audio and Asterisk error messages like
285 "replay check failed".
287 ------------------------------------------------------------------------------
288 --- New functionality introduced in Asterisk 18.0.0 --------------------------
289 ------------------------------------------------------------------------------
293 * The Streams API becomes the home for the core ACN capabilities.
296 * Parsing and formatting of codec negotiation preferences.
297 * Resolving pending streams and topologies with those configured
298 using configured preferences.
299 * Utility functions for creating string representations of
300 streams, topologies, and negotiation preferences.
302 For codec negotiation preferences:
303 * Added ast_stream_codec_prefs_parse() which takes a string
304 representation of codec negotiation preferences, which
305 may come from a pjsip endpoint for example, and populates
306 a ast_stream_codec_negotiation_prefs structure.
307 * Added ast_stream_codec_prefs_to_str() which does the reverse.
308 * Added many functions to parse individual parameter name
309 and value strings to their respective enum values, and the
313 * Added ast_stream_create_resolved() which takes a "live" stream
314 and resolves it with a configured stream and the negotiation
315 preferences to create a new stream.
316 * Added ast_stream_to_str() which create a string representation
317 of a stream suitable for debug or display purposes.
320 * Added ast_stream_topology_create_resolved() which takes a "live"
321 topology and resolves it, stream by stream, with a configured
322 topology stream and the negotiation preferences to create a new
324 * Added ast_stream_topology_to_str() which create a string
325 representation of a topology suitable for debug or display
327 * Renamed ast_format_caps_from_topology() to
328 ast_stream_topology_get_formats() to be more consistent with
329 the existing ast_stream_get_formats().
332 * A new function ast_format_cap_append_names() appends the results
333 to the ast_str buffer instead of replacing buffer contents.
337 * The BridgeAdd application now behaves more like the Bridge application.
338 The application now sets the BRIDGERESULT channel variable to indicate
339 what happened when the channel resumes in dialplan. This is instead of
340 hanging up the channel on failure conditions.
344 * Two new options, incoming_call_offer_pref and outgoing_call_offer_pref
345 have been added to res_pjsip endpoints that specify the preferred order
346 of codecs to use between those received/sent in an SDP offer and those
347 set in the endpoint configuration.
349 ------------------------------------------------------------------------------
350 --- Functionality changes from Asterisk 17.0.0 to Asterisk 18.0.0 ------------
351 ------------------------------------------------------------------------------
355 * You can now specify an optional 'Content-Type' as an argument for the Asterisk
356 SendText manager action.
360 * A new parameter 'inhibitConnectedLineUpdates' is now available in the
361 'bridges.addChannel' call. This prevents the identity of the newly connected
362 channel from being presented to other bridge members.
366 * The Channel resource has a new sub-resource "externalMedia".
367 This allows an application to create a channel for the sole purpose
368 of exchanging media with an external server. Once created, this
369 channel could be placed into a bridge with existing channels to
370 allow the external server to inject audio into the bridge or
371 receive audio from the bridge.
372 See https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI
373 for more information.
377 * H.265/HEVC is now a supported video codec and it can be used by
378 specifying "h265" in the allow line.
379 Please note however, that handling of the additional SDP parameters
380 described in RFC 7798 section 7.2 is not yet supported.
384 * Adds support for AudioSocket, a very simple bidirectional audio streaming
385 protocol. There are both channel and application interfaces.
387 A description of the protocol can be found on the referenced wiki page. A
388 short talk about the reasons and implementation can be found on YouTube at
391 ARI support has also been added via the existing "externalMedia" ARI
392 functionality. The UUID is specified using the arbitrary "data" field.
394 Wiki: https://wiki.asterisk.org/wiki/display/AST/AudioSocket
395 YouTube: https://www.youtube.com/watch?v=tjduXbZZEgI
399 * In order to reduce the amount of AMI and ARI events generated,
400 the global "Message/ast_msg_queue" channel can be set to suppress
401 it's normal channel housekeeping events such as "Newexten",
402 "VarSet", etc. This can greatly reduce load on the manager
403 and ARI applications when the Digium Phone Module for Asterisk
404 is in use. To enable, set "hide_messaging_ami_events" in
405 asterisk.conf to "yes" In Asterisk versions <18, the default
406 is "no" preserving existing behavior. Beginning with
407 Asterisk 18, the option will default to "yes".
411 * STIR/SHAKEN support has been added to Asterisk. Configuration is done in
412 stir_shaken.conf. There is a sample configuration file to help you get
413 started (asterisk/configs/samples/stir_shaken.conf.sample). Once that's
414 set up, you can enable STIR/SHAKEN on any endpoint by setting stir_shaken
415 to yes on the endpoint configuration object. This will add an Identity
416 header on outgoing INVITEs, and check for an Identity header on incoming
417 INVITEs. This option has been added to Alembic as well.
419 The information received on an incoming INVITE can be checked using the
420 STIR_SHAKEN dialplan function. There are two variations:
423 STIR_SHAKEN(0, verify_result)
425 The first variation will tell you how many STIR/SHAKEN results are on the
426 channel. The second fetches information for a specific result. The first
427 parameter is the index, followed by what information you want to retrieve.
428 The available options are 'verify_result', 'identity', and 'attestation'.
432 * The ChanIsAvail application now tolerates empty positions in the supplied
433 device list. Dialplan can now be simplified by not having to check for
434 empty positions in the device list.
438 * A new bridge profile option, maximum_sample_rate, has been added which sets
439 a maximum sample rate that the bridge will be mixed at. This allows the bridge
440 to move below the maximum sample rate as needed but caps it at the maximum.
442 * A new option, "text_messaging", has been added to the user profile
443 which allows control over whether text messaging is enabled or
444 disabled for a user. If enabled (the default) text messages
445 will be sent to the user. If disabled no text messages will be
450 * The Dial application now tolerates empty positions in the supplied
451 destination list. Dialplan can now be simplified by not having to check
452 for empty positions in the destination list. If there are no endpoints to
453 dial then DIALSTATUS is set to CHANUNAVAIL.
457 * An option 'S' has been added to MixMonitor. If used in combination with
458 the r() and/or t() options, if a frame is available to write to one of
459 those files but not the other, a frame of silence if written to the file
460 that does not have an audio frame. This should prevent the two files
461 from "drifting" when mixed after the fact.
463 * If the 'filename' argument to MixMonitor() ended with '.wav49,'
464 Asterisk would silently convert the extension to '.WAV' when opening
465 the file for writing. This caused the MIXMONITOR_FILENAME variable to
466 reference the wrong file. The MIXMONITOR_FILENAME variable will now
467 reflect the name of the file that Asterisk actually used instead of
468 the filename that was passed to the application.
472 * The Page application now tolerates empty positions in the supplied
473 destination list. Dialplan can now be simplified by not having to check
474 for empty positions in the destination list.
478 * A feature was added in Asterisk 13.27.0 and 16.4.0 that removed lock files from
479 the Asterisk voicemail directory on startup. Some users that store their
480 voicemails on network storage devices experienced slow startup times due to the
481 relative expense of traversing the voicemail directory structure looking for
482 orphaned lock files. This feature has now been removed.
484 Users who require the lock files to be removed at startup should modify their
485 startup scripts to do so before starting the asterisk process.
489 * A new dialplan function, PJSIP_MOH_PASSTHROUGH, has been added to chan_pjsip. This
490 allows the behaviour of the moh_passthrough endpoint option to be read or changed
491 in the dialplan. This allows control on a per-call basis.
495 * The UnicastRTP channel driver provided by chan_rtp now accepts
496 "<hostname>:<port>" as an alternative to "<ip_address>:<port>" in the destination.
497 The first AAAA (preferred) or A record resolved will be used as the destination.
498 The lookup is synchronous so beware of possible dialplan delays if you specify a
503 * A new parameter, httpheader, has been added to CURLOPT function. This parameter
504 allows to set custom http headers for subsequent calls off CURL function.
505 Any setting of headers will replace the default curl headers
506 (e.g. "Content-type: application/x-www-form-urlencoded")
508 * A new option, followlocation, can now be enabled with the CURLOPT()
509 dialplan function. Setting this will instruct cURL to follow 3xx
510 redirects, which it does not by default.
514 * The JITTERBUFFER dialplan function now has an option to enable video synchronization
515 support. When enabled and used with a compatible channel driver (chan_sip, chan_pjsip)
516 the video is buffered according to the size of the audio jitterbuffer and is
517 synchronized to the audio.
521 * Accept decimal number as argument.
525 * You can now disable the /httpstatus page served by Asterisk's built-in
526 HTTP server by setting 'enable_status' to 'no' in http.conf.
530 * The 'minmemfree' configuration option now counts memory allocated to
531 the filesystem cache as "free" because it is memory that is available
536 * When creating a channel in ARI using the create call
537 you can now specify dialplan variables to be set as part
538 of the same operation.
542 * This fix allows a realtime moh class to be unregistered from the command
543 line. This is useful when the contents of a directory referenced by a
544 realtime moh class have changed.
545 The realtime moh class is then reloaded on the next request and uses the
546 new directory contents.
548 * A new mode - playlist - has been added to res_musiconhold. This mode allows the
549 user to specify the files (or URLs) to play explicitly by putting them directly
554 * Added a new PJSIP system setting called disable_rport.
555 Default is no to keep support working as before.
557 If it is false (default) it adds the 'rport' parameter in the outgoing request message.
558 If it is true it does not add the 'rport' parameter in the outgoing request message.
560 This is a system option, but working as a global option.
562 res_pjsip_endpoint_identifier_ip
564 * In 'type = identify' sections, the addresses specified for the 'match'
565 clause can now include a port number. For IP addresses, the port is
566 provided by including a colon after the address, followed by the
567 desired port number. If supplied, the netmask should follow the port
568 number. To specify a port for IPv6 addresses, the address itself must
569 be enclosed in brackets to be parsed correctly.
573 * The PJSIP packet logger now has the following CLI commands:
575 pjsip set logger pcap <filename>
577 When used this will create a pcap file containing the incoming
578 and outgoing SIP packets, in unencrypted form.
580 pjsip set logger console <on / off>
582 This allows you to toggle logging to console on and off.
584 pjsip set logger host <IP/subnet mask> add
586 This allows you to add an additional IP address or subnet
587 mask to logging, allowing you to log multiple instead of
588 just a single IP address or all traffic.
590 The normal "pjsip set logger host" CLI command has also been
591 expanded to allow subnet masks as well.
595 * When placing an outgoing call to a PJSIP endpoint the intent
596 of any requested formats will now be respected. If only an audio
597 format is requested (such as ulaw) but the underlying endpoint
598 does not support the format the resulting SDP will still only
599 contain an audio stream, and not any additional streams such as
602 * Two new options, incoming_call_offer_pref and outgoing_call_offer_pref
603 have been added to res_pjsip endpoints that specify the preferred order
604 of codecs to use between those received/sent in an SDP offer and those
605 set in the endpoint configuration.
609 * This change include a new cli command 'rtp show settings'
611 The command display by general settings of rtp configuration. For this
612 point is added the fields: rtpstart, rtpend, dtmftimeout, rtpchecksum,
613 strictrtp, learning_min_sequential and icesupport.
615 * The blacklist mechanism in res_rtp_asterisk for ICE and STUN was converted to
618 As such six now options are now available:
627 These options have their obvious meanings as used elsewhere.
629 Backwards compatibility was maintained by adding {stun,ice}_blacklist as
630 aliases for {stun,ice}_deny.
632 res_sorcery_memory_cache
634 * The SorceryMemoryCacheExpireObject AMI action and CLI
635 command allow expiring of a specific object within the
636 sorcery memory cache. This is done by removing the
637 object from the cache with the expectation that the
638 cache will then re-populate the object when it is next
641 For full backend caching this does not occur. The cache
642 won't repopulate until an entire refresh is done resulting
643 in the possibility that objects are missing until that
646 The AMI action and CLI command will now not allow
647 expiring of an object if the cache is configured as a
648 full backend cache. Instead you must use either the
649 SorceryMemoryCacheExpire or SorceryMemoryCachePopulate
650 AMI actions or their associated CLI commands.
654 * Added two new CLI commands to reset stats for taskprocessors. You can
655 reset stats for a single, specific taskprocessor ('core reset
656 taskprocessor <taskprocessor>'), or you can reset all taskprocessors
657 ('core reset taskprocessors'). These commands will reset the counter for
658 the number of tasks processed as well as the max queue size.
660 * Added "like" support for 'core show taskprocessors'. Now you
661 can specify a specific set of taskprocessors (or just one) by
662 adding the keyword "like" to the above command, followed by
663 your search criteria.
665 ------------------------------------------------------------------------------
666 --- New functionality introduced in Asterisk 17.0.0 --------------------------
667 ------------------------------------------------------------------------------
671 * The bridging core no longer uses the stasis cache for bridge
672 snapshots. The latest bridge snapshot is now stored on the
673 ast_bridge structure itself.
675 The following APIs are no longer available since the stasis cache
677 ast_bridge_topic_cached()
678 ast_bridge_topic_all_cached()
680 A topic pool is now used for individual bridge topics.
682 The ast_bridge_cache() function was removed since there's no
683 longer a separate container of snapshots.
685 A new function "ast_bridges()" was created to retrieve the
686 container of all bridges. Users formerly calling
687 ast_bridge_cache() can use the new function to iterate over
688 bridges and retrieve the latest snapshot directly from the
691 The ast_bridge_snapshot_get_latest() function was renamed to
692 ast_bridge_get_snapshot_by_uniqueid().
694 A new function "ast_bridge_get_snapshot()" was created to retrieve
695 the bridge snapshot directly from the bridge structure.
697 The ast_bridge_topic_all() function now returns a normal topic
698 not a cached one so you can't use stasis cache functions on it
701 The ast_bridge_snapshot_type() stasis message now has the
702 ast_bridge_snapshot_update structure as it's data. It contains
703 the last snapshot and the new one.
707 * The core no longer uses the stasis cache for channels snapshots.
708 The following APIs are no longer available:
709 ast_channel_topic_cached()
710 ast_channel_topic_all_cached()
711 The ast_channel_cache_all() and ast_channel_cache_by_name() functions
712 now returns an ao2_container of ast_channel_snapshots rather than a
713 container of stasis_messages therefore you can't call stasis_cache
715 The ast_channel_topic_all() function now returns a normal topic,
716 not a cached one so you can't use stasis cache functions on it either.
717 The ast_channel_snapshot_type() stasis message now has the
718 ast_channel_snapshot_update structure as it's data.
719 ast_channel_snapshot_get_latest() still returns the latest snapshot.
723 * The chan_sip module is now deprecated, users should migrate to the
724 replacement module chan_pjsip. See guides at the Asterisk Wiki:
725 https://wiki.asterisk.org/wiki/x/tAHOAQ
726 https://wiki.asterisk.org/wiki/x/hYCLAQ
728 ------------------------------------------------------------------------------
729 --- Functionality changes from Asterisk 16.0.0 to Asterisk 17.0.0 ------------
730 ------------------------------------------------------------------------------
734 * A new application, this will queue up attended transfer to the given extension.
738 * A new application, this will redirect all channels currently
739 bridged to the caller channel to the specified destination.
743 * Add "average_all", "highest_all", and "lowest_all" values for
744 the remb_behavior option. These values operate on a bridge
745 level instead of a per-source level. This means that a single
746 REMB value is calculated and sent to every sender, instead of
747 a REMB value that is unique for the specific sender..
751 * Add RINGTIME and RINGTIME_MS variables containing respectively seconds and
752 milliseconds between creation of the dialing channel and receiving the first
755 Add PROGRESSTIME and PROGRESSTIME_MS variables analogous to the above with respect to
756 the PROGRESS signal. Shorter of these two times should be equivalent to
757 the PDD (Post Dial Delay) value
759 Add DIALEDTIME_MS and ANSWEREDTIME_MS variables to get millisecond resolution
760 versions of DIALEDTIME and ANSWEREDTIME
764 * You can now indicate that you'd like an ice_host_candidate's local address
765 to be published as well as the mapped address. See the sample rtp.conf
766 for more information.
770 * Add 'p' option to stop reading extension if user presses '#' key.
774 * The DUNDi PBX module now supports IPv4/IPv6 dual binding.
778 * Added a new PJSIP global setting called norefersub.
779 Default is true to keep support working as before.
781 res_pjsip_refer configures PJSIP norefersub capability accordingly.
783 Checks the PJSIP global setting value.
784 If it is true (default) it adds the norefersub capability to PJSIP.
785 If it is false (disabled) it does not add the norefersub capability
788 This is useful for Cisco switches that do not follow RFC4488.
792 * DTLS packets will now be fragmented according to the MTU as set in rtp.conf. This
793 allows larger certificates to be used for the DTLS negotiation. By default this value
796 ------------------------------------------------------------------------------
797 --- Functionality changes from Asterisk 16.2.0 to Asterisk 16.3.0 ----------
798 ------------------------------------------------------------------------------
802 * Application event filtering is now supported. An application can now specify
803 an "allowed" and/or "disallowed" list(s) of event types. Only those types
804 indicated in the "allowed" list are sent to the application. Conversely, any
805 types defined in the "disallowed" list are not sent to the application. Note
806 that if a type is specified in both lists "disallowed" takes precedence.
808 * A new REST API call has been added: 'move'. It follows the format
809 'channels/{channelId}/move' and can be used to move channels from one application
810 to another without needing to exit back into the dialplan. An application must be
811 specified, but the passing a list of arguments to the new application is optional.
812 An example call would look like this:
814 client.channels.move(channelId=chan.id, app='ari-example', appArgs='a,b,c')
816 If the channel was inside of a bridge when switching applications, it will
817 remain there. If the application specified cannot be moved to, then the channel
818 will remain in the current application and an event will be triggered named
819 "ApplicationMoveFailed", which will provide the destination application's name
820 and the channel information.
824 * A new configuration parameter "taskprocessor_overload_trigger" has been
825 added to the pjsip.conf "globals" section. The distributor currently stops
826 accepting new requests when any taskprocessor overload is triggered. The
827 new option allows you to completely disable overload detection (NOT
828 RECOMMENDED), keep the current behavior, or trigger only on pjsip
829 taskprocessor overloads.
833 * A new configuration parameter 'ignore_183_without_sdp' has been added
834 to the pjsip.conf "endpoints" section. If enabled, will make chan_pjsip
835 discard 183s that do not contain an SDP body, which can resolve no
836 ringback tone issues as well as making the behavior match chan_sip.
840 * A new module "res_mwi_devstate" has been added that allows subscriptions
841 to voicemail boxes using "presence" events. This allows common BLF keys
842 to act as voicemail waiting indicators.
846 * Added the ability to set the wrapuptime per-member using the AddQueueMember
849 ------------------------------------------------------------------------------
850 --- Functionality changes from Asterisk 16.1.0 to Asterisk 16.2.0 ------------
851 ------------------------------------------------------------------------------
855 * Whenever an ARI application is started, a context will be created for it
856 automatically as long as one does not already exist, following the format
857 'stasis-<app_name>'. Two extensions are also added to this context: a match-all
858 extension, and the 'h' extension. Any phone that registers under this context
859 will place all calls to the corresponding Stasis application.
863 * Added "send_contact_status_on_update_registration" global configuration option
864 to enable sending AMI ContactStatus event when a device refreshes its registration.
868 * Reworked the media indexer so it doesn't cache the index. Testing revealed
869 that the cache added no benefit but that it could consume excessive memory.
870 Two new index related functions were created: ast_sounds_get_index_for_file()
871 and ast_media_index_update_for_file() which restrict index updating to
872 specific sound files. The original ast_sounds_get_index() and
873 ast_media_index_update() calls are still available but since they no longer
874 cache the results internally, developers should re-use an index they may
875 already have instead of calling ast_sounds_get_index() repeatedly. If
876 information for only a single file is needed, ast_sounds_get_index_for_file()
877 should be called instead of ast_sounds_get_index().
881 * Before Asterisk 12, when using the automon or automixmon features defined
882 in features.conf, a channel variable (TOUCH_MIXMONITOR_OUTPUT) was set on
883 both channels, indicating the filename of the recording.
885 When bridging was overhauled in Asterisk 12, the behavior was changed such
886 that the variable was only set on the peer channel and not on the channel
887 that initiated the automon or automixmon.
889 The previous behavior has been restored so both channels receive the
890 channel variable when one of these features is invoked.
894 * You can now specify a special context with the "aliasescontext" parameter
895 in voicemail.conf which will allow you to create aliases for physical
898 ------------------------------------------------------------------------------
899 --- Functionality changes from Asterisk 16.0.0 to Asterisk 16.1.0 ------------
900 ------------------------------------------------------------------------------
904 * pbx_config will now find and process multiple 'globals' sections from
905 extensions.conf. Variables are processed in the order they are found
906 and duplicate variables overwrite the previous value.
910 * New dialplan function PJSIP_PARSE_URI added to parse an URI and return
911 a specified part of the URI.
915 * ast_bt_get_symbols() now returns a vector of strings instead of an
916 array of strings. This must be freed with ast_bt_free_symbols.
920 * New options 'trust_connected_line' and 'send_connected_line' have been
921 added to the endpoint. The option 'trust_connected_line' is to control
922 if connected line updates are accepted from this endpoint.
923 The option 'send_connected_line' is to control if connected line updates
924 can be sent to this endpoint.
925 The default value is 'yes' for both options.
929 * The existing strictrtp option in rtp.conf has a new choice availabe, called
930 'seqno', which behaves the same way as setting strictrtp to 'yes', but will
931 ignore the time interval during learning so that bursts of packets can still
932 trigger learning our source.
934 ------------------------------------------------------------------------------
935 --- Functionality changes from Asterisk 15 to Asterisk 16 --------------------
936 ------------------------------------------------------------------------------
940 * The app_fax module is now deprecated, users should migrate to the
941 replacement module res_fax.
945 * An 'a' option has been added to the Originate dialplan application which
946 will execute the originate in an asynchronous fashion. If set then the
947 application will return immediately without waiting for the originated
952 * MALLOC_DEBUG no longer has an effect on Asterisk's ABI. Asterisk built
953 with MALLOC_DEBUG can now successfully load binary modules built without
954 MALLOC_DEBUG and vice versa. Third-party pre-compiled modules no longer
955 need to have a special build with it enabled.
957 * Asterisk now depends on libjansson >= 2.11. If this version is not
958 available on your distro you can use `./configure --with-jansson-bundled`.
962 * The app_macro module is now deprecated and by default it is no longer
963 built. Users should migrate to app_stack (Gosub). A warning is logged
964 the first time any Macro is used.
968 * The app_setcallerid module has been removed. The CALLERID dialplan function
969 should be used instead.
973 * New function SIP_HEADERS() enumerates all headers in the incoming INVITE.
975 * The variable GET_TRANSFERRER_DATA set in the peer channel causes matching
976 headers be retrieved from the REFER message and made accessible to the
977 dialplan in the hash TRANSFER_DATA.
981 * Timeouts for reading digits from analog phones are now configurable in
982 chan_dahdi.conf: firstdigit_timeout, interdigit_timeout, matchdigit_timeout.
986 * The ContactStatus and Status fields for the manager events ContactStatus
987 and ContactStatusDetail are now set to "NonQualified" when a contact exists
988 but has not been qualified.
990 * The "Newexten" event is now part of the "dialplan" class. The documentation
991 for Asterisk 15 already specified this, but the implementation was actually
992 using the "call" class instead.
996 * The ContactInfo event's contact_status field is now set to "NonQualified"
997 when a contact exists but has not been qualified.
1001 * Added the ability to set the wrapuptime in the configuration of member.
1002 When set the wrapuptime on the member is used instead of the wrapuptime
1003 defined for the queue itself.
1005 * Added predial handler support for caller and callee channels with the
1006 B and b options respectively. This is similar to the predial support
1011 * The res_config_sqlite module is now deprecated, users should migrate to the
1012 replacement module res_config_sqlite3.
1016 * The res_monitor module is now deprecated, users should migrate to the
1017 replacement module app_mixmonitor.
1021 * A new AMI action, PJSIPShowAors, has been added which displays information
1022 about all configured PJSIP AORs.
1024 * A new AMI action, PJSIPShowAuths, has been added which displays information
1025 about all configured PJSIP Auths.
1027 * A new AMI action, PJSIPShowContacts, has been added which displays information
1028 about all configured PJSIP Contacts.
1030 res_pjsip_registrar_expire
1032 * The res_pjsip_registrar_expire module has been removed. The functionality has
1033 been moved into res_pjsip_registrar.
1035 func_audiohookinherit
1037 * The func_audiohookinherit module has been removed. Due to architectural changes
1038 in Asterisk 12, audiohook inheritance is performed automatically and this
1039 function now lacks function.
1043 * The cdr_syslog module is now deprecated and by default it is no longer
1048 * The cdr_sqlite module has been removed. Users should move to using the
1049 cdr_sqlite3_custom module instead.
1053 * The format_jpeg module has been removed.
1057 * DUNDi now supports IPv6
1061 * libedit is no longer available as an embedded library and must be provided
1063 * The STATIC_BUILD functionality has been removed as it has not been maintained
1064 and has not worked in quite some time.
1065 * The module loader now enforces inter-module dependencies. This ensures that
1066 a module is not started before another it depends on, even if preload is used.
1067 If a dependency is not available or fails to startup this will block any
1068 dependants from startup.
1069 * Parts of the Asterisk core which can load configuration from realtime are now
1070 built-in modules. It is no longer necessary to preload realtime drivers as
1071 they are always initialized before the built-in modules.
1073 ------------------------------------------------------------------------------
1074 --- Functionality changes from Asterisk 15.5.0 to Asterisk 15.6.0 ------------
1075 ------------------------------------------------------------------------------
1079 * A new option 'suppress_q850_reason_headers' has been added to the endpoint
1080 object. Some devices can't accept multiple Reason headers and get confused
1081 when both 'SIP' and 'Q.850' Reason headers are received. This option allows
1082 the 'Q.850' Reason header to be suppressed. The default value is 'no'.
1084 res_pjsip_endpoint_identifier_ip
1086 * Added regex support to the identify section match_header option. You
1087 specify a regex instead of an explicit string by surrounding the header
1089 match_header = SIPHeader: /regex/
1091 ------------------------------------------------------------------------------
1092 --- Functionality changes from Asterisk 15.4.0 to Asterisk 15.5.0 ------------
1093 ------------------------------------------------------------------------------
1097 * Core bridging and, more specifically, bridge_softmix have been enhanced to
1098 relay received frames of type TEXT or TEXT_DATA to all participants in a
1099 softmix bridge. res_pjsip_messaging and chan_pjsip have been enhanced to
1100 take advantage of this so when res_pjsip_messaging receives an in-dialog
1101 MESSAGE message from a user in a conference call, it's relayed to all
1102 other participants in the call.
1106 * Support Enhanced Messaging. SendText now accepts new channel variables
1107 that can be used to override the To and From display names and set the
1108 Content-Type of a message. Since you can now set Content-Type, other
1109 text/* content types are now valid.
1113 * ConfbridgeList now shows talking status. This utilizes the same voice
1114 detection as the ConfbridgeTalking event, so bridges must be configured
1115 with "talk_detection_events=yes" for this flag to have meaning.
1117 * ConfBridge can now send events to participants via in-dialog MESSAGEs.
1118 All current Confbridge events are supported, such as ConfbridgeJoin,
1119 ConfbridgeLeave, etc. In addition to those events, a new event
1120 ConfbridgeWelcome has been added that will send a list of all
1121 current participants to a new participant.
1125 * Two new options have been added to the system and endpoint objects to
1126 control whether, on outbound calls, Asterisk will accept updated SDP answers
1127 during the initial INVITE transaction when 100rel is not in effect.
1128 This usually happens when the INVITE is forked to multiple UASs and more
1129 than one sends an SDP answer or when a single UAS needs to change a media
1130 port to switch from custom ringback to the actual media destination.
1132 The 'follow_early_media_forked' option sets whether Asterisk will accept
1133 the updated SDP when the To tag on the subsequent response is different than
1134 that on the the previous response. This usually occurs in the forked INVITE
1135 scenario. The default value is "yes" which is the current behavior.
1137 The 'accept_multiple_sdp_answers' flag sets whether Asterisk will accept the
1138 updated SDP when the To tag on the subsequent response is the same as that
1139 on the previous response. This can occur when a UAS needs to switch media
1140 ports from custom ringback to the final media path. The default value is
1141 "no" which is the current behavior.
1143 These options have to be enabled system-wide in the system config section
1144 of pjsip.conf as well as on individual endpoints that require the
1147 ------------------------------------------------------------------------------
1148 --- Functionality changes from Asterisk 15.3.0 to Asterisk 15.4.0 ------------
1149 ------------------------------------------------------------------------------
1153 * A new configuration option "genericplc_on_equal_codecs" was added to the
1154 "plc" section of codecs.conf to allow generic packet loss concealment even
1155 if no transcoding was originally needed. Transcoding via SLIN is forced
1160 * Added the "cache_pools" option to pjproject.conf. Disabling the option
1161 helps track down pool content mismanagement when using valgrind or
1162 MALLOC_DEBUG. The cache gets in the way of determining if the pool contents
1163 are used after free and who freed it.
1167 * Extend the PJSIPNotify AMI command to send an in-dialog notify on a
1170 ------------------------------------------------------------------------------
1171 --- Functionality changes from Asterisk 15.2.0 to Asterisk 15.3.0 ------------
1172 ------------------------------------------------------------------------------
1176 * During dialplan reload log messages are produced for each context,
1177 extension and include. These messages are no longer printed by the
1178 verbose loggers, they are now only logged as debug messages.
1182 * Added the Muted header to the ConfbridgeJoin AMI event to indicate the
1183 participant's starting mute status.
1185 * Made the AMI ConfbridgeList action's ConfbridgeList events output all
1186 the standard channel snapshot headers instead of a few hand-coded channel
1187 snapshot headers. The benefit is that the CallerIDName gets disruptive
1188 characters like CR, LF, Tab, and a few others escaped. However, an empty
1189 CallerIDName is now output as "<unknown>" instead of "<no name>".
1193 * Added a new prompt, connecting-prompt, which will be played
1194 (if configured) to the "winner" callee before connecting the call.
1198 * Users who are matching endpoints by SIP header need to reevaluate their
1199 global "endpoint_identifier_order" option in light of the "ip" endpoint
1200 identifier method split into the "ip" and "header" endpoint identifier
1203 * The pjsip_transport_event feature introduced in 15.1.0 has been refactored.
1204 Any external modules that may have used that feature (highly unlikely) will
1205 need to be changed as the API has been altered slightly.
1207 res_pjsip_endpoint_identifier_ip
1209 * The endpoint identifier "ip" method previously recognized endpoints either
1210 by IP address or a matching SIP header. The "ip" endpoint identifier method
1211 is now split into the "ip" and "header" endpoint identifier methods. The
1212 "ip" endpoint identifier method only matches by IP address and the "header"
1213 endpoint identifier method only matches by SIP header. The split allows the
1214 user to control the relative priority of the IP address and the SIP header
1215 identification methods in the global "endpoint_identifier_order" option.
1216 e.g., If you have two type=identify sections where one matches by IP address
1217 for endpoint alice and the other matches by SIP header for endpoint bob then
1218 you can now predict which endpoint is matched when a request comes in that
1223 * In an earlier release, inbound registrations on a reliable transport
1224 were pruned on Asterisk restart since the TCP connection would have
1225 been torn down and become unusable when Asterisk stopped. This same
1226 process is now also applied to inbound subscriptions. Since this
1227 required the addition of a new column to the ps_subscription_persistence
1228 realtime table, users who store their subscriptions in a database will
1229 need to run the "alembic upgrade head" process to add the column to
1232 res_pjsip_transport_management
1234 * Since res_pjsip_transport_management provides several attack
1235 mitigation features, its functionality moved to res_pjsip and
1236 this module has been removed. This way the features will always
1237 be available if res_pjsip is loaded.
1239 ------------------------------------------------------------------------------
1240 --- Functionality changes from Asterisk 15.1.0 to Asterisk 15.2.0 ------------
1241 ------------------------------------------------------------------------------
1245 * Added the "cache_media_frames" option to asterisk.conf. Disabling the option
1246 helps track down media frame mismanagement when using valgrind or
1247 MALLOC_DEBUG. The cache gets in the way of determining if the frame is
1248 used after free and who freed it. NOTE: This option has no effect when
1249 Asterisk is compiled with the LOW_MEMORY compile time option enabled because
1250 the cache code does not exist.
1254 * Calls to invalid extensions are now reported as an ACL failure security event
1255 "no_extension_match".
1259 * The X.509 certificate used for DTLS negotiation can now be automatically
1260 generated. This is supported by res_pjsip by specifying
1261 "dtls_auto_generate_cert = yes" on a PJSIP endpoint. For chan_sip, you
1262 would set "dtlsautogeneratecert = yes" either in the [general] section of
1263 sip.conf or on a specific peer.
1267 * The "identify_by" on endpoints can now be set to "ip" to restrict an endpoint
1268 being matched based only on IP address. To ensure no behavior change the
1269 default has been changed to "username,ip".
1271 ------------------------------------------------------------------------------
1272 --- Functionality changes from Asterisk 15.0.0 to Asterisk 15.1.0 ------------
1273 ------------------------------------------------------------------------------
1277 * The "remove_existing" option now allows a registration to succeed by
1278 displacing any existing contacts that now exceed the "max_contacts" count.
1279 Any removed contacts are the next to expire. The behaviour change is
1280 beneficial when "rewrite_contact" is enabled and "max_contacts" is greater
1281 than one. The removed contact is likely the old contact created by
1282 "rewrite_contact" that the device is refreshing.
1286 * Added a new CancelAtxfer action that cancels an attended transfer.
1288 ------------------------------------------------------------------------------
1289 --- Functionality changes from Asterisk 14 to Asterisk 15 --------------------
1290 ------------------------------------------------------------------------------
1294 * PAUSEALL/UNPAUSEALL now sets the pause reason in the queue_log if it has
1297 * A new option, "announce-position-only-up," has been added that, when set to
1298 yes, causes position announcements to only be played when the caller's
1299 queue position has improved since the last time that we announced their
1300 position. This default is no.
1304 * '--with-pjproject-bundled' is now the default when running ./configure
1305 It can be disabled with '--without-pjproject-bundled'.
1307 * A '--with-download-cache' option is now available which is equivalent to
1308 setting '--with-sounds-cache' and '--with-externals-cache' to the same
1309 value. The download cache can also be set via the AST_DOWNLOAD_CACHE
1310 environment variable.
1312 ------------------------------------------------------------------------------
1313 --- Functionality changes from Asterisk 14.6.0 to Asterisk 14.7.0 ------------
1314 ------------------------------------------------------------------------------
1318 * The "external_media_address" on transports is now resolved using dnsmgr and
1319 when dnsmgr refreshes are enabled will be automatically updated with the new
1320 IP address of a given hostname.
1322 * A new endpoint parameter "incoming_mwi_mailbox" allows Asterisk to receive
1323 unsolicited MWI NOTIFY requests and make them available to other modules via
1324 the stasis message bus.
1328 * By default, when res_musiconhold reloads or unloads, it sends a HUP signal
1329 to custom applications (and all descendants), waits 100ms, then sends a
1330 TERM signal, waits 100ms, then finally sends a KILL signal. An application
1331 which is interacting with an external device and/or spawns children of its
1332 own may not be able to exit cleanly in the default times, expecially if sent
1333 a KILL signal, or if it's children are getting signals directly from
1334 res_musiconhoild. To allow extra time, the 'kill_escalation_delay'
1335 class option can be used to set the number of milliseconds res_musiconhold
1336 waits before escalating kill signals, with the default being the current
1337 100ms. To control to whom the signals are sent, the "kill_method"
1338 class option can be set to "process_group" (the default, existing behavior),
1339 which sends signals to the application and its descendants directly, or
1340 "process" which sends signals only to the application itself.
1342 * New dialplan function PJSIP_DTMF_MODE added to get or change the DTMF mode
1343 of a channel on a per-call basis.
1347 * OAuth 2.0 authentication is now supported when contacting Google. Follow the
1348 instructions in xmpp.conf.sample to retrieve and configure the necessary
1351 ------------------------------------------------------------------------------
1352 --- Functionality changes from Asterisk 14.5.0 to Asterisk 14.6.0 ------------
1353 ------------------------------------------------------------------------------
1357 * A new global option "imap_poll_logout" was added to specify whether need to
1358 disconnect from the IMAP server after polling of mailboxes.
1363 * A new endpoint option "refer_blind_progress" was added to turn off notifying
1364 the progress details on Blind Transfer. If this option is not set then
1365 the chan_pjsip will send NOTIFY "200 OK" immediately after "202 Accepted".
1366 On default is enabled.
1367 Some SIP phones like Mitel/Aastra or Snom keep the line busy until
1370 * A new endpoint option "notify_early_inuse_ringing" was added to control
1371 whether to notify dialog-info state 'early' or 'confirmed' on Ringing
1374 * The endpoint option 'dtmf_mode' has a new option 'auto_dtmf' added. This
1375 mode works similar to 'auto' except uses DTMF INFO as fallback instead of
1380 * The EAGI() application will now look for a dialplan variable named
1381 EAGI_AUDIO_FORMAT and use that format with the 'enhanced' audio pipe that
1382 EAGI provides. If not specified, it will continue to use the default signed
1387 * When dialing an endpoint directly or using the PJSIP_DIAL_CONTACTS dialplan
1388 function any contact which is considered unreachable due to qualify being
1389 enabled will no longer be called.
1391 * The asymmetric_rtp_codec option now also controls whether chan_pjsip will
1392 send media as-is without transcoding if the codec has been negotiated in the
1393 SDP. If set to "no" then Asterisk will only ever send the preferred codec
1394 from the SDP, unless the remote side sends a different codec and we will
1399 * Added a new PJPROJECT_CONFIGURE_OPTS environment variable which can be used
1400 to pass arbitrary options to the bundled pjproject configure.
1402 * Automatically set the bundled pjproject configure --host and --build
1403 options to match those supplied for the asterisk configure.
1405 ------------------------------------------------------------------------------
1406 --- Functionality changes from Asterisk 14.4.0 to Asterisk 14.5.0 ------------
1407 ------------------------------------------------------------------------------
1411 * Added the stun_blacklist option to rtp.conf. Some multihomed servers have
1412 IP interfaces that cannot reach the STUN server specified by stunaddr.
1413 Blacklist those interface subnets from trying to send a STUN packet to find
1414 the external IP address. Attempting to send the STUN packet needlessly
1415 delays processing incoming and outgoing SIP INVITEs because we will wait
1416 for a response that can never come until we give up on the response.
1417 Multiple subnets may be listed.
1421 * Added logger_queue_limit to the configuration options.
1422 All log messages go to a queue serviced by a single thread
1423 which does all the IO. This setting controls how big that
1424 queue can get (and therefore how much memory is allocated)
1425 before new messages are discarded.
1426 The default is 1000.
1428 res_pjsip_config_wizard
1430 * Two new parameters have been added to the pjsip config wizard.
1431 Setting 'sends_line_with_registrations' to true will cause the wizard
1432 to skip the creation of an identify object to match incoming requests
1433 to the endpoint and instead add the line and endpoint parameters to
1434 the outbound registration object.
1435 Setting 'outbound_proxy' is a shortcut for adding individual
1436 endpoint/outbound_proxy, aor/outbound_proxy and registration/outbound_proxy
1441 * If the 'call-id' value is specified for the uuid_type option and a
1442 chan_sip channel is used the resulting HEP traffic will now contain the
1443 SIP Call-ID instead of the Asterisk channel name.
1445 ------------------------------------------------------------------------------
1446 --- Functionality changes from Asterisk 14.3.0 to Asterisk 14.4.0 ------------
1447 ------------------------------------------------------------------------------
1451 * LOW_MEMORY no longer has an effect on Asterisk ABI. Symbols that were
1452 previously suppressed by LOW_MEMORY are now replaced by stub functions.
1453 Asterisk built with LOW_MEMORY can now successfully load binary modules
1454 built without LOW_MEMORY and vice versa.
1456 * RADIUS backends for CEL and CDR can now also be built using the radcli
1457 client library, in addition to the existing support for building them
1458 using either freeradius or radiusclient-ng.
1462 * ASTERISK_REGISTER_FILE was no longer useful and has been removed. Sources
1463 which use mtx_prof must now manually declare and initialize the variable.
1467 * If an offer is received with optional SRTP (a media stream with RTP/AVP but
1468 which contains a crypto line) chan_sip will now accept it and enable SRTP.
1469 If you would like to do optional SRTP on outbound you will need to create
1470 a dialplan that dials with it enabled initially and if it fails fall back to
1475 * Added endpoint configuration parameter "preferred_codec_only".
1476 This allow asterisk response to a SIP invite with the single most
1477 preferred codec rather than advertising all joint codec capabilities.
1478 This limits the other side's codec choice to exactly what we prefer.
1482 * To fix a memory leak the syslog channel is now empty if it has not been set
1483 and used by a syslog channel in the logger.
1487 * To fix a memory leak the syslog channel is now empty if it has not been set
1488 and used by a syslog channel in the logger.
1492 * New setting "rtp_pt_dynamic = 35" in asterisk.conf:
1493 Normally the Dynamic RTP Payload Type numbers are 96-127, which allow just 32
1494 formats. To avoid the message "No Dynamic RTP mapping available", the range
1495 was changed to 35-63,96-127. This is allowed by RFC 3551 section 3. However,
1496 when you use more than 32 formats and calls are not accepted by a remote
1497 implementation, please report this and go back to rtp_pt_dynamic = 96.
1499 * A new setting, "rtp_use_dynamic", has been added in asterisk.conf". When set
1500 to "yes" RTP dynamic payload types are assigned dynamically per RTP instance.
1501 When set to "no" RTP dynamic payload types are globally initialized to pre-
1502 designated numbers and function similar to static payload types.
1506 * Added support to gosub predial routines on both original channel and on the
1507 created channel using options parameter (like app_dial) B() and b(). This
1508 allows for adding variables to newly created channel or, e.g. setting callerid.
1512 * 'dialplan show' output will now show [config_file:line_number] instead of
1513 [registrar] when that information is available. Currently only extensions
1514 registered by pbx_config when loading/reloading will use this format.
1518 * Add 'QueueUpdate' application which can be used to track outbound calls
1523 * Asterisk will now set the AST_OUTGOING_ATTEMPT channel variable so that
1524 attempt-specific behavior is possible. This is a 1-based number that
1525 simply increases by 1 for each attempt.
1527 ------------------------------------------------------------------------------
1528 --- Functionality changes from Asterisk 14.3.0 to Asterisk 14.4.0 ------------
1529 ------------------------------------------------------------------------------
1533 * The 'PJSIPShowEndpoint' command's respone event of 'IdentifyDetail' now
1534 contains a new optional parameter, 'MatchHeader', mapping to the new
1535 configuration option 'match_header' for the corresponding 'identify' object.
1536 It should be noted that since 'match_header' takes in a key: value pair, the
1537 event parameter will contain a ':' as well.
1541 * Added new 'u' option to Record() application which prevents Asterisk from
1542 truncating silence from the end of recorded files.
1544 res_pjsip_outbound_registration
1546 * Outbound registrations are now refreshed when res_stun_monitor detects
1547 a network change event has happened.
1548 The 'pjsip send (un)register' CLI commands were updated to accept '*all'
1549 as an argument to operate on all registrations.
1550 The 'PJSIP(Un)Register' AMI commands were updated to also accept '*all'.
1554 * The 'Comedian Mail' prompts can now be overriden using the 'vm-login' and
1555 'vm-newuser' configuration options in voicemail.conf.
1557 * Added 'fromstring' field to the voicemail boxes. If set, it will override
1558 the global 'fromstring' field on a per-mailbox basis.
1562 * Added CHANNEL(callid) to retrieve the call log tag associated with the
1563 channel. e.g., [C-00000000] Dialplan now has access to the call log
1564 search key associated with the channel so it can be saved in case there
1565 is a problem with the call.
1569 * A new transport parameter 'symmetric_transport' has been added.
1570 When a request from a dynamic contact comes in on a transport with this
1571 option set to 'yes', the transport name will be saved and used for
1572 subsequent outgoing requests like OPTIONS, NOTIFY and INVITE. It's
1573 saved as a contact uri parameter named 'x-ast-txp' and will display with
1574 the contact uri in CLI, AMI, and ARI output. On the outgoing request,
1575 if a transport wasn't explicitly set on the endpoint AND the request URI
1576 is not a hostname, the saved transport will be used and the 'x-ast-txp'
1577 parameter stripped from the outgoing packet. To facilitate recreation of
1578 subscriptions on asterisk restart, a new column 'contact_uri' needed to be
1579 added to the ps_subcsription_persistence table. Since new columns were
1580 added to both transport and subscription_persistence, an alembic upgrade
1581 should be run to bring the database tables up to date.
1583 * A new option, allow_overlap, has been added to endpoints which allows
1584 overlap dialing functionality to be enabled or disabled. The option defaults
1587 res_pjsip_transport_websocket
1589 * Removed non-secure websocket support. Firefox and Chrome have not allowed
1590 non-secure websockets for quite some time so this shouldn't be an issue
1591 for people. Attempting to use a non-secure websocket may or may not work
1592 when Asterisk attempts to send SIP requests to do something like initiate
1595 res_pjsip_endpoint_identifier_ip
1597 * A new option has been added to the 'identify' configuration object,
1598 'match_header'. The 'match_header' attribute should contain a SIP
1599 header: value pair that, When set, will cause inbound requests that contain
1600 the matching SIP header/value pair to be associated with the corresponding
1601 endpoint. This option is cumulative with the 'match' option, so that if
1602 either option matches the request, the request is associated with the
1605 In a future release, this module will be renamed to something more
1606 appropriate, as it now matches inbound requests on more than just IP
1611 * The RTP layer of Asterisk now has support for RFC 5761: "Multiplexing RTP
1612 Data and Control Packets on a Single Port." So far, the only channel driver
1613 that supports this feature is chan_pjsip. You can set "rtcp_mux = yes" on
1614 a PJSIP endpoint in pjsip.conf to enable the feature.
1616 ------------------------------------------------------------------------------
1617 --- Functionality changes from Asterisk 14.2.0 to Asterisk 14.3.0 ------------
1618 ------------------------------------------------------------------------------
1622 * Added new CLI command "pjproject set log level". The new command allows
1623 the maximum PJPROJECT log levels to be adjusted dynamically and
1624 independently from the set debug logging level like many other similar
1625 module debug logging commands.
1627 * Added new companion CLI command "pjproject show log level" to allow the
1628 user to see the current maximum pjproject logging level.
1630 * Added new pjproject.conf startup section "log_level' option to set the
1631 initial maximum PJPROJECT logging level.
1633 res_pjsip_outbound_registration
1635 * Statsd no longer logs redundant status PJSIP.registrations.state changes
1636 for internal state transitions that don't change the reported public status
1641 * The PJSIPShowRegistrationInboundContactStatuses AMI command has been added
1642 to return ContactStatusDetail events as opposed to
1643 PJSIPShowRegistrationsInbound which just a dumps every defined AOR.
1647 * Six existing contact fields have been added to the end of the
1648 ContactStatusDetail AMI event:
1649 ID, AuthenticateQualify, OutboundProxy, Path, QualifyFrequency and
1650 QualifyTimeout. Existing fields have not been disturbed.
1652 res_pjsip_endpoint_identifier_ip
1654 * SRV lookups can now be done on provided hostnames to determine additional
1655 source IP addresses for requests. This is configurable using the
1656 "srv_lookups" option on the identify and defaults to "yes".
1660 * The 'ari set debug' command has been enhanced to accept 'all' as an
1661 application name. This allows dumping of all apps even if an app
1662 hasn't registered yet.
1664 * 'ari set debug' now displays requests and responses as well as events.
1666 ------------------------------------------------------------------------------
1667 --- Functionality changes from Asterisk 14.1.0 to Asterisk 14.2.0 ------------
1668 ------------------------------------------------------------------------------
1672 * Events that reference a bridge may now contain two new optional fields:
1673 - 'BridgeVideoSourceMode': the video source mode for the bridge.
1674 Can be one of 'none', 'talker', or 'single'.
1675 - 'BridgeVideoSource': the unique ID of the channel that is the video
1676 source in this bridge, if one exists.
1678 * A new event, BridgeVideoSourceUpdate, has been added with a class
1679 authorization of CALL. The event is raised when the video source changes
1680 in a multi-party mixing bridge.
1684 * The bridges resource now exposes two new operations:
1685 - POST /bridges/{bridgeId}/videoSource/{channelId}: Set a video source in a
1686 multi-party mixing bridge
1687 - DELETE /bridges/{bridgeId}/videoSource: Remove the set video source,
1688 reverting to talk detection for the video source
1690 * The bridge model in any returned response or event now contains the following
1692 - video_mode: the video source mode for the bridge. Can be one of 'none',
1693 'talker', or 'single'.
1694 - video_source_id: the unique ID of the channel that is the video source
1695 in this bridge, if one exists.
1697 * A new event, BridgeVideoSourceChanged, has been added for bridges.
1698 Applications subscribed to a bridge will receive this event when the source
1699 of video changes in a mixing bridge.
1701 * The ARI major version has been bumped. There are not any known breaking changes
1702 in ARI. The major version has been bumped because otherwise we can end up with
1703 overlapping version numbers between different Asterisk versions. Now each major
1704 version of Asterisk will bring with it a change in the major version of ARI.
1705 The ARI version in Asterisk 14 is now 2.0.0.
1709 * Automatic dual stack support is now implemented. Depending on DNS resolution
1710 and the transport used for sending a message the SIP signaling and SDP will
1711 be updated with the correct IP address and protocol version. This means that
1712 the rtp_ipv6 and t38_udptl_ipv6 options no longer have any effect. The
1713 res_pjsip_multihomed module has also been moved into core res_pjsip to ensure
1714 that messages are updated with the correct address information in all cases.
1718 * The default behavior for RTP codecs has been changed. The sending codec will
1719 now match the receiving codec. This can be turned off and behavior reverted
1720 to asymmetric using the "asymmetric_rtp_codec" endpoint option. If this
1721 option is set then the sending and received codec are allowed to differ.
1725 * Three new CLI commands have been added for ARI:
1727 Displays a listing of all registered ARI applications.
1728 - ari show app <name>:
1729 Display detailed information about a registered ARI application.
1730 - ari set debug <name> <on|off>:
1731 Enable/disable debugging of an ARI application. When debugged, verbose
1732 information will be sent to the Asterisk CLI.
1737 * A new dialplan variable, ABANDONED, is set when the call is not answered
1742 * The configuration file ari.conf now supports a channelvars option, which
1743 specifies a list of channel variables to include in each channel-oriented
1746 ------------------------------------------------------------------------------
1747 --- Functionality changes from Asterisk 14.0.0 to Asterisk 14.1.0 ------------
1748 ------------------------------------------------------------------------------
1752 * The res_digium_phone, codec_g729a, codec_silk, codec_siren7 and
1753 codec_siren14 binary modules hosted at downloads.digium.com can now be
1754 automatically downloaded and installed during the Asterisk install
1755 process. If selected in menuselect, when 'make install' is run, the
1756 script will check the downloads site for a new version and download
1757 and install it if needed. The '--with-externals-cache' option to
1758 ./configure can be used to specify a location to cache the latest
1759 tarballs so they don't have to be re-downloaded for every install.
1763 * Added "tps_queue_high" and "tps_queue_low" options.
1764 The options can modify the taskprocessor alert levels for this module.
1765 Additional information can be found in the sample configuration file at
1766 config/samples/voicemail.conf.sample.
1770 * Added "mwi_tps_queue_high" and "mwi_tps_queue_low" global configuration
1771 options to tune taskprocessor alert levels.
1773 * Added "mwi_disable_initial_unsolicited" global configuration option
1774 to disable sending unsolicited MWI to all endpoints on startup.
1775 Additional information can be found in the sample configuration file at
1776 config/samples/pjsip.conf.sample.
1780 * A new dialplan function, PJSIP_SEND_SESSION_REFRESH, has been added. When
1781 invoked, a re-INVITE or UPDATE request will be sent immediately to the
1782 endpoint underlying the channel. When used in combination with the existing
1783 dialplan function PJSIP_MEDIA_OFFER, this allows the formats on a PJSIP
1784 channel to be re-negotiated and updated after session set up.
1788 * A new endpoint configuration parameter 'contact_user' has been added which
1789 when set will override the default user set on Contact headers in outgoing
1792 * If you are using a sorcery realtime backend to store global res_pjsip
1793 options (ps_globals table) then you now have to do a res_pjsip reload for
1794 changes to these options to take effect. If you are using pjsip.conf to
1795 configure these options then you already had to do a reload after making
1798 * Added "ignore_uri_user_options" global configuration option for
1799 compatibility with an ITSP that sends URI user field options. When enabled
1800 the user field is truncated at the first semicolon.
1802 URI: "sip:1235557890;phone-context=national@x.x.x.x;user=phone"
1803 The user field is "1235557890;phone-context=national"
1804 Which is truncated to this: "1235557890"
1806 Note: The caller-id and redirecting number strings obtained from incoming
1807 SIP URI user fields are now always truncated at the first semicolon.
1811 * An option, ice_blacklist, has been added which allows certain subnets to be
1812 excluded from local ICE candidates.
1816 * Some sounds played into the bridge are played asynchronously. This, for
1817 instance, allows a channel to immediately exit the ConfBridge without having
1818 to wait for a leave announcement to play.
1822 * Added the "Q" option which sets the Q.850/Q.931 cause on unanswered channels
1823 when another channel answers the call. The default of ANSWERED_ELSEWHERE
1828 * ARI events will all now include a new field in the root of the JSON message,
1829 'asterisk_id'. This will be the unique ID for the Asterisk system
1830 transmitting the event. The value can be overridden using the 'entityid'
1831 setting in asterisk.conf.
1833 ------------------------------------------------------------------------------
1834 --- Functionality changes from Asterisk 13 to Asterisk 14 --------------------
1835 ------------------------------------------------------------------------------
1839 * A new event, "DialState" has been added. This is similar to "DialBegin" and
1840 "DialEnd" in that it tracks the state of a dialed call. The difference is that
1841 this indicates some intermediate state change in the dial attempt, such as
1842 "RINGING", "PROGRESS", or "PROCEEDING".
1846 * A new ARI method has been added to the channels resource. "create" allows for
1847 you to create a new channel and place that channel into a Stasis application.
1848 This is similar to origination except that the specified channel is not
1849 dialed. This allows for an application writer to create a channel, perform
1850 manipulations on it, and then delay dialing the channel until later.
1852 * To complement the "create" method, a "dial" method has been added to the
1853 channels resource in order to place a call to a created channel.
1855 * All operations that initiate playback of media on a resource now support
1856 a list of media URIs. The list of URIs are played in the order they are
1857 presented to the resource. A new event, "PlaybackContinuing", is raised when
1858 a media URI finishes but before the next media URI starts. When a list is
1859 played, the "Playback" model will contain the optional attribute
1860 "next_media_uri", which specifies the next media URI in the list to be played
1861 back to the resource. The "PlaybackFinished" event is raised when all media
1864 * Stored recordings now allow for the media associated with a stored recording
1865 to be retrieved. The new route, GET /recordings/stored/{name}/file, will
1866 transmit the raw media file to the requester as binary.
1869 * "Dial" events have been modified to not only be sent when dialing begins and ends.
1870 They now are also sent for intermediate states, such as "RINGING", "PROGRESS", and
1878 * A new application in Asterisk, this will join the calling channel
1879 to an existing bridge containing the named channel prefix.
1883 * Added the 'l' option, which forces ChanSpy's audiohook to use a long queue
1884 to store the audio frames. This option is useful if audio loss is
1885 experienced when using ChanSpy, but may introduce some delay in the audio
1886 feed on the listening channel.
1890 * Added format attribute negotiation for the iLBC audio codec. Format attribute
1891 negotiation is provided by the res_format_attr_ilbc module. iLBC 20 is the
1892 default now. Falls back to iLBC 30, when the remote party requests this.
1896 * Added the ability to pass options to MixMonitor when recording is used with
1897 ConfBridge. This includes the addition of the following configuration
1898 parameters for the 'bridge' object:
1899 - record_file_timestamp: whether or not to append the start time to the
1901 - record_options: the options to pass to the MixMonitor application
1902 - record_command: a command to execute when recording is finished
1903 Note that these options may also be with the CONFBRIDGE function.
1907 * Remote files can now be retrieved and played back. See the Playback
1908 dialplan application for more details.
1912 * It is now possible to disable the prompt from a callee by setting
1913 'enable_callee_prompt = no' in followme.conf.
1917 * Remote files can now be retrieved and played back via the Playback and other
1918 media playback dialplan applications. This is done by directly providing
1919 the URL to play to the dialplan application:
1920 same => n,Playback(http://1.1.1.1/howler-monkeys-fl.wav)
1921 Note that unlike 'normal' media files, the entire URI to the file must be
1922 provided, including the file extension. Currently, on HTTP and HTTPS URI
1923 schemes are supported.
1927 * Added field ReasonPause on QueueMemberStatus if set when paused, the reason
1928 the queue member was paused.
1930 * Added field LastPause on QueueMemberStatus for time when started the last
1931 pause for a queue member.
1933 * Show the time when started the last pause for queue member on CLI for command
1938 * Added the 'n' option, which prevents the SMS from being written to the log
1939 file. This is needed for those countries with privacy laws that require
1940 providers to not log SMS content.
1948 * The CALLERID(ani2) value for incoming calls is now populated in featdmf
1949 signaling mode. The information was previously discarded.
1951 * Added the force_restart_unavailable_chans compatibility option. When
1952 enabled it causes Asterisk to restart the ISDN B channel if an outgoing
1953 call receives cause 44 (Requested channel not available).
1957 * The iax.conf forcejitterbuffer option has been removed. It is now always
1958 forced if you set iax.conf jitterbuffer=yes. If you put a jitter buffer
1959 on a channel it will be on the channel.
1961 * A new configuration parameters, 'calltokenexpiration', has been added that
1962 controls the duration before a call token expires. Default duration is 10
1963 seconds. Setting this to a higher value may help in lagged networks or those
1964 experiencing high packet loss.
1966 * Plaintext auth mode is deprecated and removed from possible default modes.
1968 chan_rtp (was chan_multicast_rtp)
1970 * Added unicast RTP support and renamed chan_multicast_rtp to chan_rtp.
1972 * The format for dialing a unicast RTP channel is:
1973 UnicastRTP/<destination-addr>[/[<options>]]
1974 Where <destination-addr> is something like '127.0.0.1:5060'.
1975 Where <options> are in standard Asterisk flag options format:
1976 c(<codec>) - Specify which codec/format to use such as 'ulaw'.
1977 e(<engine>) - Specify which RTP engine to use such as 'asterisk'.
1979 * New options were added for a multicast RTP channel. The format for
1980 dialing a multicast RTP channel is:
1981 MulticastRTP/<type>/<destination-addr>[/[<control-addr>][/[<options>]]]
1982 Where <type> can be either 'basic' or 'linksys'.
1983 Where <destination-addr> is something like '224.0.0.3:5060'.
1984 Where <control-addr> is something like '127.0.0.1:5060'.
1985 Where <options> are in standard Asterisk flag options format:
1986 c(<codec>) - Specify which codec/format to use such as 'ulaw'.
1987 i(<address>) - Specify the interface address from which multicast RTP
1989 l(<enable>) - Set whether packets are looped back to the sender. The
1990 enable value can be 0 to set looping to off and non-zero to set
1992 t(<ttl>) - Set the time-to-live (TTL) value for multicast packets.
1996 * New 'rtpbindaddr' global setting. This allows a user to define which
1997 ipaddress to bind the rtpengine to. For example, chan_sip might bind
1998 to eth0 (10.0.0.2) but rtpengine to eth1 (192.168.1.10).
2000 * DTLS related configuration options can now be set at a general level.
2001 Enabling DTLS support, though, requires enabling it at the user
2004 * Added the possibility to set the From: header through the the SIP dial
2005 string (populating the fromuser/fromdomain fields), complementing the
2006 [!dnid] option for the To: header that has existed since 1.6.0 (1d6b192).
2007 NOTE: This is again separated by an exclamation mark, so the To: header may
2008 not contain one of those.
2010 * Session-Timers (RFC 4028) work for TCP (and TLS) transports as well now.
2011 Previously Asterisk dropped calls only with UDP transports. However with
2012 longer international calls via TCP, the SIP channel might break, because
2013 all hops on the Internet route must stay online (have not a single power
2014 outage, for example). Therefore with Session-Timers enabled (which are
2015 enabled at default), you might see additional dropped calls. Consequently
2016 please, consider to go for session-timers=refuse in your sip.conf.
2020 * New 'user_eq_phone' endpoint setting. This adds a 'user=phone' parameter
2021 to the request URI and From URI if the user is determined to be a phone
2024 * New 'moh_passthrough' endpoint setting. This will pass hold and unhold
2025 requests through using SIP re-invites with sendonly and sendrecv accordingly.
2027 * Added the pjsip.conf system type disable_tcp_switch option. The option
2028 allows the user to disable switching from UDP to TCP transports described
2029 by RFC 3261 section 18.1.1.
2031 * New 'line' and 'endpoint' options added on outbound registrations. This
2032 allows some identifying information to be added to the Contact of the
2033 outbound registration. If this information is present on messages received
2034 from the remote server the message will automatically be associated with the
2035 configured endpoint on the outbound registration.
2040 * The core of Asterisk uses a message bus called "Stasis" to distribute
2041 information to internal components. For performance reasons, the message
2042 distribution was modified to make use of a thread pool instead of a
2043 dedicated thread per consumer in certain cases. The initial settings for
2044 the thread pool can now be configured in 'stasis.conf'.
2046 * A new core DNS API has been implemented which provides a common interface
2047 for DNS functionality. Modules that use this functionality will require that
2048 a DNS resolver module is loaded and available.
2050 * Modified processing of command-line options to first parse only what
2051 is necessary to read asterisk.conf. Once asterisk.conf is fully loaded,
2052 the remaining options are processed. The -X option now applies to
2053 asterisk.conf only. To enable #exec for other config files you must
2054 set execincludes=yes in asterisk.conf. Any other option set on the
2055 command-line will now override the equivalent setting from asterisk.conf.
2057 * The TLS core in Asterisk now supports X.509 certificate subject alternative
2058 names. This way one X.509 certificate can be used for hosts that can be
2059 reached under multiple DNS names or for multiple hosts.
2061 * The Asterisk logging system now supports JSON structured logging. Log
2062 channels specified in logger.conf or added dynamically via CLI commands now
2063 support an optional specifier prior to their levels that determines their
2064 formatting. To set a log channel to format its entries as JSON, a formatter
2065 of '[json]' can be set, e.g.,
2066 full => [json]debug,verbose,notice,warning,error
2068 * The core now supports a 'media cache', which stores temporary media files
2069 retrieved from external sources. CLI commands have been added to manipulate
2070 and display the cached files, including:
2071 - 'media cache show <all>' - show all cached media files, or details about
2072 one particular cached media file
2073 - 'media cache refresh <item>' - force a refresh of a particular media file
2075 - 'media cache delete <item>' - remove an item from the cache
2076 - 'media cache create <uri>' - retrieve a URI and store it in the cache
2078 * The ability for device state hints to be automatically created as a result of
2079 device state changes now exists in the PBX. This functionality is referred to
2080 as "autohints" and is configurable in extensions.conf by placing "autohints=yes"
2081 in the context. If enabled a device state hint will be automatically created
2082 with the name of the device.
2084 * If Asterisk is built with systemd support, and run under systemd, it will
2085 notify systemd of its state using sd_notify. Use 'Type=notify' in
2090 * The func_odbc global option "single_db_connection" default value has been
2096 * New module format_ogg_speex added which supports Speex codec inside
2097 Ogg containers (filename extension .spx).
2102 * Added CHANNEL(onhold) item that returns 1 (onhold) and 0 (not-onhold) for
2103 the hold status of a channel.
2107 * The CURL function now supports a write option, which will save the retrieved
2108 file to a location on disk. As an example:
2109 same => n,Set(CURL(https://1.1.1.1/foo.wav)=/tmp/foo.wav)
2110 will save 'foo.wav' to /tmp.
2114 * The transferdialattempts default value has been changed from 1 to 3. The
2115 transferinvalidsound has been changed from "pbx-invalid" to
2116 "privacy-incorrect". These were changed to make DTMF transfers be more
2117 user-friendly by default.
2123 res_http_media_cache
2125 * A backend for the core media cache, this module retrieves media files from
2126 a remote HTTP(S) server and stores them in the core media cache for later
2131 * Added sort=randstart to the sort options. It sorts the files by name and
2132 then chooses the first file to play at random.
2133 * Added preferchannelclass=no option to prefer the application-passed class
2134 over the channel-set musicclass. This allows separate hold-music from
2135 application (e.g. Queue or Dial) specified music.
2137 res_resolver_unbound
2139 * Added a res_resolver_unbound module which uses the libunbound resolver library
2140 to perform DNS resolution. This module requires the libunbound library to be
2141 installed in order to be used.
2145 * A new SIP resolver using the core DNS API has been implemented. This relies on
2146 external SIP resolver support in PJSIP which is only available as of PJSIP
2147 2.4. If this support is unavailable the existing built-in PJSIP SIP resolver
2148 will be used instead. The new SIP resolver provides NAPTR support, improved
2149 SRV support, and AAAA record support.
2151 res_pjsip_info_empty
2152 --------------------
2153 * A new module that can respond to empty Content-Type INFO packets during call.
2154 Some SBCs will terminate a call if their empty INFO packets are not responded
2155 to within a predefined time.
2157 res_pjsip_outbound_registration
2158 -------------------------------
2159 * A new 'fatal_retry_interval' option has been added to outbound registration.
2160 When set (default is zero), and upon receiving a failure response to an
2161 outbound registration, registration is retried at the given interval up to
2164 res_pjsip_outbound_publish
2166 * Added a new multi_user option that when set to 'yes' allows a given configuration
2167 to be used for multiple users.
2175 * Added a new option, 'usegmtime', which causes timestamps in CEL events
2176 to be logged in GMT.
2178 * Added support to set schema where located the table cel. This settings is
2179 configurable for cel_pgsql via the 'schema' in configuration file
2188 * Added the ability to set the character to quote identifiers. This
2189 allows adding the character at the start and end of table and column
2190 names. This setting is configurable for cdr_adaptive_odbc via the
2191 quoted_identifiers in configuration file cdr_adaptive_odbc.conf.
2195 * Added a new configuration option, "newcdrcolumns", which enables use of the
2196 post-1.8 CDR columns 'peeraccount', 'linkedid', and 'sequence'.
2200 * Added a new configuration option, "newcdrcolumns", which enables use of the
2201 post-1.8 CDR columns 'peeraccount', 'linkedid', and 'sequence'.
2203 ------------------------------------------------------------------------------
2204 --- Functionality changes from Asterisk 13.10.0 to Asterisk 13.11.0 ----------
2205 ------------------------------------------------------------------------------
2209 * Added "faxdetect_timeout" option.
2210 The option determines how many seconds into a call before faxdetect
2211 is disabled for the call. Setting the value to zero disables the timeout.
2215 * Added "fax_detect_timeout" to endpoint.
2216 The option determines how many seconds into a call before fax_detect
2217 is disabled for the call. Setting the value to zero disables the timeout.
2219 * Added "subscribe_context" to endpoint.
2220 If specified, incoming SUBSCRIBE requests will be searched for the matching
2221 extension in the indicated context. If no "subscribe_context" is specified,
2222 then the "context" setting is used.
2226 * The DTLS part in Asterisk now supports Perfect Forward Secrecy (PFS).
2227 Enabling PFS is attempted by default, and is dependent on the configuration
2228 of the module using TLS.
2229 - Ephemeral ECDH (ECDHE) is enabled by default. To disable it, do not
2230 specify a ECDHE cipher suite in sip.conf, for example:
2231 dtlscipher=AES128-SHA
2232 - Ephemeral DH (DHE) is disabled by default. To enable it, add DH parameters
2233 into the private key file, e.g., sip.conf dtlsprivatekey. For example:
2234 openssl dhparam -out ./dh.pem 2048
2235 - Because clients expect the server to prefer PFS, and because OpenSSL sorts
2236 its cipher suites by bit strength, see "openssl ciphers -v DEFAULT".
2237 Consider re-ordering your cipher suites in the respective configuration
2239 dtlscipher=ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-RSA-AES128-GCM-SHA256
2240 which forces PFS and requires at least DTLS 1.2.
2242 ------------------------------------------------------------------------------
2243 --- Functionality changes from Asterisk 13.9.0 to Asterisk 13.10.0 -----------
2244 ------------------------------------------------------------------------------
2248 * A channel variable FORWARDERNAME is now set which indicates which channel
2249 was responsible for a forwarding requests received on dial attempt.
2253 * Added new global option "single_db_connection".
2254 Enabling this option func_odbc will use a single database connection per DSN.
2255 This option is enabled by default.
2259 * Added FAXMODE variable to let dialplan know what fax transport was used.
2260 FAXMODE variable is set to either "audio" or "T38".
2264 * Added "via_addr", "via_port", "call_id" to contacts.
2265 As res_pjsip_nat rewrites contact's address, only the last Via header
2266 can contain the source address of registered endpoint.
2267 Also Call-Id header may contain the source address of registered endpoint.
2268 Added new fields ViaAddress,CallID to AMI event ContactStatus
2270 * Endpoint IP Access Controls
2271 Added new configuration Endpoint options:
2272 "acl" - list of IP ACL section names in acl.conf
2273 "deny" - List of IP addresses to deny access from
2274 "permit" - List of IP addresses to permit access from
2275 "contact_acl" - List of Contact ACL section names in acl.conf
2276 "contact_deny" - List of Contact header addresses to deny
2277 "contact_permit" - List of Contact header addresses to permit
2279 * Added "reg_server" to contacts.
2280 If the Asterisk system name is set in asterisk.conf, it will be stored
2281 into the "reg_server" field in the ps_contacts table to facilitate
2282 multi-server setups.
2284 * When starting Asterisk, received traffic will now be ignored until Asterisk
2285 has loaded all modules and is fully booted.
2289 * Added a new option, 'uuid_type', that sets the preferred source of the Homer
2290 correlation UUID. The valid options are:
2291 - call-id: Use the PJSIP SIP Call-ID header value
2292 - channel: Use the Asterisk channel name
2293 The default value is 'call-id'. In the event that a HEP module cannot find a
2294 valid value using the specified 'uuid_type', the module may fallback to a
2295 more readily available source for the correlation UUID.
2299 * A new option has been added, 'max_connections', which sets the maximum number
2300 of concurrent connections to the database. This option defaults to 1 which
2301 returns the behavior to that of Asterisk 13.7 and prior.
2305 * Added a bridge profile option called regcontext that allows you to
2306 dynamically register the conference bridge name as an extension into
2307 the specified context. This allows tracking down conferences on multi-
2308 server installations via alternate means (DUNDI for example). By default
2309 this feature is not used.
2313 * Added the associated format name to 'core show codecs'.
2317 * Added 'formats' to channel create/originate to allow setting the allowed
2318 formats for a channel when no originator channel is available. Especially
2319 useful for Local channel creation where no other format information is
2320 available. 'core show codecs' can now be used to look up suitable format
2323 ------------------------------------------------------------------------------
2324 --- Functionality changes from Asterisk 13.8.0 to Asterisk 13.9.0 ------------
2325 ------------------------------------------------------------------------------
2328 - The dynamic parking lot creation channel variables PARKINGDYNAMIC,
2329 PARKINGDYNCONTEXT, PARKINGDYNEXTEN, and PARKINGDYNPOS are now looked
2330 for in the parker's channel instead of the parked channel. This is only
2331 of significance if the parker uses blind transfer or the DTMF one-step
2332 parking feature. You need to use the double underscore '__' inheritance
2333 for these variables. The indefinite inheritance is also recommended
2334 for the PARKINGEXTEN variable.
2338 * Added new global option (disable_multi_domain) to pjsip.
2339 Disabling Multi Domain can improve realtime performace by reducing
2340 number of database requsts.
2344 * Added 'pjsip show channelstats' CLI command.
2346 res_pjsip_outbound_publish
2348 * Added support for setting the transport used on outbound publish
2349 using the transport configuration option.
2351 ------------------------------------------------------------------------------
2352 --- Functionality changes from Asterisk 13.7.0 to Asterisk 13.8.0 ------------
2353 ------------------------------------------------------------------------------
2357 * Per RFC3325, the 'From' header is now anonymized on outgoing calls when
2358 caller id presentation is prohibited.
2360 res_pjsip_config_wizard
2362 * A new command (pjsip export config_wizard primitives) has been added that
2363 will export all the pjsip objects it created to the console or a file
2364 suitable for reuse in a pjsip.conf file.
2368 * To help insure that Asterisk is compiled and run with the same known
2369 version of pjproject, a new option (--with-pjproject-bundled) has been
2370 added to ./configure. When specified, the version of pjproject specified
2371 in third-party/versions.mak will be downloaded and configured. When you
2372 make Asterisk, the build process will also automatically build pjproject
2373 and Asterisk will be statically linked to it. Once a particular version
2374 of pjproject is configured and built, it won't be configured or built
2375 again unless you run a 'make distclean'.
2377 To facilitate testing, when 'make install' is run, the pjsua and pjsystest
2378 utilities and the pjproject python bindings will be installed in
2379 ASTDATADIR/third-party/pjproject.
2381 The default behavior remains building with the shared pjproject
2382 installation, if any.
2386 * Added CONFBRIDGE_INFO(muted,) for querying the muted conference state.
2388 * Added Muted header to AMI ConfbridgeListRooms action response list events
2389 to indicate the muted conference state.
2391 * Added Muted column to CLI "confbridge list" output to indicate the muted
2392 conference state and made the locked column a yes/no value instead of a
2393 locked/unlocked value.
2397 * The REDIRECTING(reason) value is now treated consistently between
2398 chan_sip and chan_pjsip.
2400 Both channel drivers match incoming reason values with values documented
2401 by REDIRECTING(reason) and values documented by RFC5806 regardless of
2402 whether they are quoted or not. RFC5806 values are mapped to the
2403 equivalent REDIRECTING(reason) documented value and is set in
2404 REDIRECTING(reason). e.g., an incoming RFC5806 'unconditional' value or a
2405 quoted string version ('"unconditional"') is converted to
2406 REDIRECTING(reason)'s 'cfu' value. The user's dialplan only needs to deal
2407 with 'cfu' instead of any of the aliases.
2409 The incoming 480 response reason text supported by chan_sip checks for
2410 known reason values and if not matched then puts quotes around the reason
2411 string and assigns that to REDIRECTING(reason).
2413 Both channel drivers send outgoing known REDIRECTING(reason) values as the
2414 unquoted RFC5806 equivalent. User custom values are either sent as is or
2415 with added quotes if SIP doesn't allow a character within the value as
2416 part of a RFC3261 Section 25.1 token. Note that there are still
2417 limitations on what characters can be put in a custom user value. e.g.,
2418 embedding quotes in the middle of the reason string is just going to cause
2421 * Setting a REDIRECTING(reason) value now recognizes RFC5806 aliases.
2422 e.g., Setting REDIRECTING(reason) to 'unconditional' is converted to the
2427 * This module is the successor of res_pjsip_log_forwarder. As well as
2428 handling the log forwarding (which now displays as 'pjproject:0' instead
2429 of 'pjsip:0'), it also adds a 'pjproject show buildopts' command to the CLI.
2430 This displays the compiled-in options of the pjproject installation
2431 Asterisk is currently running against.
2433 * Another feature of this module is the ability to map pjproject log levels
2434 to Asterisk log levels, or to suppress the pjproject log messages
2435 altogether. Many of the messages emitted by pjproject itself are the result
2436 of errors which Asterisk will ultimately handle so the messages can be
2437 misleading or just noise. A new config file (pjproject.conf) has been added
2438 to configure the mapping and a new CLI command (pjproject show log mappings)
2439 has been added to display the mappings currently in use.
2443 * Transports are now reloadable. In testing, no in-progress calls were
2444 disrupted if the ip address or port weren't changed, but the possibility
2445 still exists. To make sure there are no unintentional drops, a new option
2446 'allow_reload', which defaults to 'no' has been added to transport. If
2447 left at the default, changes to the particular transport will be ignored.
2448 If set to 'yes', changes (if any) will be applied.
2450 * Added new global option (regcontext) to pjsip. When set, Asterisk will
2451 dynamically create and destroy a NoOp priority 1 extension
2452 for a given endpoint who registers or unregisters with us.
2454 * Endpoints and aors can now be identified by the username and realm in an
2455 incoming Authorization header. To use this feature, add "auth_username"
2456 to your endpoint's "identify_by" list. You can combine "auth_username"
2457 and the original "username" to test both the From/To and Authorization
2458 headers. For endpoints, the order is controlled by the global
2459 "endpoint_identifier_order" setting. For matching aors to an endpoint
2460 for inbound registration, the order is controlled by this option.
2462 * In conjunction with the "auth_username" change, 3 new options have been
2463 added to the global configuration object that control how many unidentified
2464 requests over a certain period from the same IP address can be received
2465 before a security alert is generated. A new CLI command
2466 "pjsip show unidentified_requests" will list the current candidates.
2470 * A new module, res_pjsip_history, has been added that provides SIP history
2471 viewing/filtering from the CLI. The module is intended to be used on systems
2472 with busy SIP traffic, where existing forms of viewing SIP messages - such
2473 as the res_pjsip_logger - may be inadequate. The module provides two new
2475 - 'pjsip set history {on|off|clear}' - this enables/disables SIP history
2476 capturing, as well as clears an existing history capture. Note that SIP
2477 packets captured are stored in memory until cleared. As a result, the
2478 history capture should only be used for debugging/viewing purposes, and
2479 should *NOT* be left permanently enabled on a system.
2480 - 'pjsip show history' - displays the captured SIP history. When invoked
2481 with no options, the entire captured history is displayed. Two options
2483 -- 'entry <num>' - display a detailed view of a single SIP message in
2485 -- 'where ...' - filter the history based on some expression. For more
2486 information on filtering, view the current CLI help for the
2487 'pjsip show history' command.
2491 * app_voicemail and res_mwi_external can now be built together. The default
2492 remains to build app_voicemail and not res_mwi_external but if they are
2493 both built, the load order will cause res_mwi_external to load first and
2494 app_voicemail will be skipped. Use 'preload=app_voicemail.so' in
2495 modules.conf to force app_voicemail to be the voicemail provider.
2499 * A new option (bind_rtp_to_media_address) has been added to endpoint which
2500 will cause res_pjsip_sdp_rtp to actually bind the RTP instance to the
2501 media_address as well as using it in the SDP. If set, RTP packets will now
2502 originate from the media address instead of the operating system's "primary"
2507 * A new configuration section - ice_host_candidates - has been added to
2508 rtp.conf, allowing automatically discovered ICE host candidates to be
2509 overriden. This allows an Asterisk server behind a 1:1 NAT to send its
2510 external IP as a host candidate rather than relying on STUN to discover it.
2512 ------------------------------------------------------------------------------
2513 --- Functionality changes from Asterisk 13.6.0 to Asterisk 13.7.0 ------------
2514 ------------------------------------------------------------------------------
2518 * Added format attribute negotiation for the VP8 video codec. Format attribute
2519 negotiation is provided by the res_format_attr_vp8 module.
2523 * A new "timeout" user profile option has been added. This configures the number
2524 of seconds that a participant may stay in the ConfBridge after joining. When
2525 the time expires, the user is ejected from the conference and CONFBRIDGE_RESULT
2526 is set to "TIMEOUT" on the channel.
2530 * The websockets_enabled option has been added to the general section of
2531 sip.conf. The option is enabled by default to match the previous behavior.
2532 The option should be disabled when using res_pjsip_transport_websockets to
2533 ensure chan_sip will not conflict with PJSIP websockets.
2537 * The HOLD_INTERCEPT dialplan function now actually exists in the source tree.
2538 While support for the events was added in Asterisk 13.4.0, the function
2539 accidentally never made it in. That function is now present, and will cause
2540 the 'hold' raised by a channel to be intercepted and converted into an
2543 res_pjsip_outbound_registration
2544 -------------------------------
2545 * If res_statsd is loaded and a StatsD server is configured, basic statistics
2546 regarding the state of outbound registrations will now be emitted. This
2548 - A GAUGE statistic for the overall number of outbound registrations, i.e.:
2549 PJSIP.registrations.count
2550 - A GAUGE statistic for the overall number of outbound registrations in a
2551 particular state, e.g.:
2552 PJSIP.registrations.state.Registered
2556 * The ability to use "like" has been added to the pjsip list and show
2557 CLI commands. For instance: CLI> pjsip list endpoints like abc
2559 * If res_statsd is loaded and a StatsD server is configured, basic statistics
2560 regarding the state of PJSIP contacts will now be emitted. This includes:
2561 - A GAUGE statistic for the overall number of contacts in a particular
2563 PJSIP.contacts.states.Reachable
2564 - A TIMER statistic for the RTT time for each qualified contact, e.g.:
2565 PJSIP.contacts.alice@@127.0.0.1:5061.rtt
2567 res_sorcery_memory_cache
2568 ------------------------
2569 * A new caching strategy, full_backend_cache, has been added which caches
2570 all stored objects in the backend. When enabled all objects will be
2571 expired or go stale according to the configuration. As well when enabled
2572 all retrieval operations will be performed against the cache instead of
2577 * CALLERID(pres) is now documented as a valid alternative to setting both
2578 CALLERID(name-pres) and CALLERID(num-pres) at once. Some channel drivers,
2579 like chan_sip, don't make a distinction between the two: they take the
2580 least public value from name-pres and num-pres. By using CALLERID(pres)
2581 for reading and writing, you touch the same combined value in the dialplan.
2582 The same applies to CONNECTEDLINE(pres), REDIRECTING(orig-pres),
2583 REDIRECTING(to-pres) and REDIRECTING(from-pres).
2587 * A new module that emits StatsD statistics regarding Asterisk endpoints.
2588 This includes a total count of the number of endpoints, the count of the
2589 number of endpoints in the technology agnostic state of the endpoint -
2590 online or offline - as well as the number of channels associated with each
2591 endpoint. These are recorded as three different GAUGE statistics:
2593 - endpoints.state.{unknown|offline|online}
2594 - endpoints.{tech}.{resource}.channels
2597 ------------------------------------------------------------------------------
2598 --- Functionality changes from Asterisk 13.5.0 to Asterisk 13.6.0 ------------
2599 ------------------------------------------------------------------------------
2603 * The CHANNEL function, when used on a PJSIP channel, now exposes a 'call-id'
2604 extraction option when using with the 'pjsip' signalling option. It will
2605 return the SIP Call-ID associated with the INVITE request that established
2610 * Two new endpoint related events are now available: PeerStatusChange and
2611 ContactStatusChange. In particular, these events are useful when subscribing
2612 to all event sources, as they provide additional endpoint related
2613 information beyond the addition/removal of channels from an endpoint.
2615 * Added the ability to subscribe to all ARI events in Asterisk, regardless
2616 of whether the application 'controls' the resource. This is useful for
2617 scenarios where an ARI application merely wants to observe the system,
2618 as opposed to control it. There are two ways to accomplish this:
2619 (1) Via the WebSocket connection URI. A new query paramter, 'subscribeAll',
2620 has been added that, when present and True, will subscribe all
2621 specified applications to all ARI event sources in Asterisk.
2622 (2) Via the applications resource. An ARI client can, at any time, subscribe
2623 to all resources in an event source merely by not providing an explicit
2624 resource. For example, subscribing to an event source of 'channels:'
2625 as opposed to 'channels:12345' will subscribe the application to all
2628 ------------------------------------------------------------------------------
2629 --- Functionality changes from Asterisk 13.4.0 to Asterisk 13.5.0 ------------
2630 ------------------------------------------------------------------------------
2634 * A new ContactStatus event has been added that reflects res_pjsip contact
2635 lifecycle changes: Created, Removed, Reachable, Unreachable, Unknown.
2637 * Added the Linkedid header to the common channel headers listed for each
2638 channel in AMI events.
2642 * A new feature has been added that enables the retrieval of modules and
2643 module information through an HTTP request. Information on a single module
2644 can be also be retrieved. Individual modules can be loaded to Asterisk, as
2645 well as unloaded and reloaded.
2647 * A new resource has been added to the 'asterisk' resource, 'config/dynamic'.
2648 This resource allows for push configuration of sorcery derived objects
2649 within Asterisk. The resource supports creation, retrieval, updating, and
2650 deletion. Sorcery derived objects that are manipulated by this resource
2651 must have a sorcery wizard that supports the desired operations.
2653 * A new feature has been added that allows for the rotation of log channels
2654 through HTTP requests.
2659 * A new 'g726_non_standard' endpoint option has been added that, when set to
2660 'yes' and g.726 audio is negotiated, forces the codec to be treated as if it
2661 is AAL2 packed on the channel.
2663 * A new 'rtp_keepalive' endpoint option has been added. This option specifies
2664 an interval, in seconds, at which we will send RTP comfort noise packets to
2665 the endpoint. This functions identically to chan_sip's "rtpkeepalive" option.
2667 * New 'rtp_timeout' and 'rtp_timeout_hold' endpoint options have been added.
2668 These options specify the amount of time, in seconds, that Asterisk will wait
2669 before terminating the call due to lack of received RTP. These are identical
2670 to chan_sip's rtptimeout and rtpholdtimeout options.
2672 ------------------------------------------------------------------------------
2673 --- Functionality changes from Asterisk 13.3.0 to Asterisk 13.4.0 ------------
2674 ------------------------------------------------------------------------------
2678 * New 'rpid_immediate' option to control if connected line update information
2679 goes to the caller immediately or waits for another reason to send the
2680 connected line information update. See the online option documentation for
2681 more information. Defaults to 'no' as setting it to 'yes' can result in
2682 many unnecessary messages being sent to the caller.
2684 * The configuration setting 'progressinband' now defaults to 'no', which
2685 matches the actual behavior of previous versions.
2689 * A new CLI command has been added: "pjsip show settings", which shows
2690 both the global and system configuration settings.
2692 * A new aor option has been added: "qualify_timeout", which sets the timeout
2693 in seconds for a qualify. The default is 3 seconds. This overrides the
2694 hard coded 32 seconds in pjproject.
2696 * Endpoint status will now change to "Unreachable" when all contacts are
2697 unavailable. When any contact becomes available, the endpoint will status
2698 will change back to "Reachable".
2700 * A new global option has been added: "max_initial_qualify_time", which
2701 sets the maximum amount of time from startup that qualifies should be
2702 attempted on all contacts.
2706 * Two new events, 'ChannelHold' and 'ChannelUnhold', have been added to the
2707 events data model. These events are raised when a channel indicates a hold
2708 or unhold, respectively.
2712 * A new dialplan function, HOLD_INTERCEPT, has been added. This function, when
2713 placed on a channel, intercepts hold/unhold indications signalled by the
2714 channel and prevents them from moving on to other channels in a bridge with
2715 the hold initiator. Instead, AMI or ARI events are raised indicating that
2716 the channel wanted to place someone on hold. This allows external
2717 applications to implement their own custom hold/unhold logic.
2719 ------------------------------------------------------------------------------
2720 --- Functionality changes from Asterisk 13.2.0 to Asterisk 13.3.0 ------------
2721 ------------------------------------------------------------------------------
2723 chan_pjsip/app_transfer
2725 * The Transfer application, when used with chan_pjsip, now supports using
2726 a PJSIP endpoint as the transfer destination. This is in addition to
2727 explicitly specifying a SIP URI to transfer to.
2731 * The ARI /channels resource now supports a new operation, 'redirect'. The
2732 redirect operation will perform a technology and state specific redirection
2733 on the channel to a specified endpoint or destination. In the case of SIP
2734 technologies, this is either a 302 Redirect response to an on-going INVITE
2735 dialog or a SIP REFER request.
2739 * A new 'endpoint_identifier_order' option has been added that allows one to
2740 set the order by which endpoint identifiers are processed and checked. This
2741 option is specified under the 'global' type configuration section.
2743 ------------------------------------------------------------------------------
2744 --- Functionality changes from Asterisk 13.1.0 to Asterisk 13.2.0 ------------
2745 ------------------------------------------------------------------------------
2747 * New 'PJSIP_AOR' and 'PJSIP_CONTACT' dialplan functions have been added which
2748 allow examining PJSIP AORs or contacts from the dialplan.
2750 res_pjsip_outbound_registration
2752 * The 'pjsip send unregister' command now stops further registrations.
2754 * A new command 'pjsip send register' has been added which allows you to
2755 start or restart periodic registration. It can be used after a
2756 'send unregister' or after a 401 permanent error.
2758 res_pjsip_config_wizard
2760 * This is a new module that adds streamlined configuration capability for
2761 chan_pjsip. It's targeted at users who have lots of basic configuration
2762 scenarios like 'phone' or 'agent' or 'trunk'. Additional information
2763 can be found in the sample configuration file at
2764 config/samples/pjsip_wizard.conf.sample.
2768 * The T.38 negotiation timeout was previously hard coded at 5000 milliseconds
2769 and is now configurable via the 't38timeout' configuration option in
2770 res_fax.conf and via the fax options dialplan function 'FAXOPT(t38timeout)'.
2771 The default remains at 5000 milliseconds.
2775 * The ca_list_path transport parameter has been added for TLS transports. This
2776 option behaves similarly to the old sip.conf option "tlscapath". In order to
2777 use this, you must be using PJProject version 2.4 or higher.
2781 * The Originate operation now takes in an originator channel. The linked ID of
2782 this originator channel is applied to the newly originated outgoing channel.
2783 If using CEL this allows an association to be established between the two so
2784 it can be recognized that the originator is dialing the originated channel.
2786 * "language" (the default spoken language for the channel) is now included in
2787 the standard channel state output for suitable events.
2789 * The POST channels/{id} operation and the POST channels/{id}/continue operation
2790 now have a new "label" parameter. This allows for origination or continuation
2791 to a labeled priority in the dialplan instead of requiring a specific priority
2792 number. The ARI version has been bumped to 1.7.0 as a result.
2796 * "Language" (the default spoken language for the channel) is now included in
2797 the standard channel state output for suitable events.
2799 * AMI actions that return a list of events have been made to return consistent
2800 headers for the action response event starting the list and the list complete
2801 event. The AMI version has been bumped to 2.7.0 as a result.
2803 ------------------------------------------------------------------------------
2804 --- Functionality changes from Asterisk 13.0.0 to Asterisk 13.1.0 ------------
2805 ------------------------------------------------------------------------------
2809 * Event NewConnectedLine is emitted when the connected line information on
2814 * Event ChannelConnectedLine is emitted when the connected line information
2815 on a channel changes.
2820 The features.conf general section has three new configurable options:
2821 * transferdialattempts
2822 * transferretrysound
2823 * transferinvalidsound
2824 For more information on what these options do, see the Asterisk wiki:
2825 https://wiki.asterisk.org/wiki/x/W4fAAQ
2832 * New 'media_encryption_optimistic' endpoint setting. This will use SRTP
2833 when possible but does not consider lack of it a failure.
2835 res_pjsip_endpoint_identifer_ip
2837 * New CLI commands have been added: "pjsip show identif(y|ies)", which lists
2838 all configured PJSIP identify objects
2840 ------------------------------------------------------------------------------
2841 --- Functionality changes from Asterisk 12 to Asterisk 13 --------------------
2842 ------------------------------------------------------------------------------
2847 Asterisk 13 is the next Long Term Support (LTS) release of Asterisk. As such,
2848 the focus of development for this release of Asterisk was on improving the
2849 usability and features developed in the previous Standard release, Asterisk 12.
2850 Beyond a general refinement of end user features, development focussed heavily
2851 on the Asterisk APIs - the Asterisk Manager Interface (AMI) and the Asterisk
2852 REST Interface (ARI) - and the PJSIP stack in Asterisk. Some highlights of the
2853 new features include:
2855 * Asterisk security events are now provided via AMI, allowing end users to
2856 monitor their Asterisk system in real time for security related issues.
2857 * External control of Message Waiting Indicators (MWI) through both AMI and ARI.
2858 * Reception/transmission of out of call text messages using any supported
2859 channel driver/protocol stack through ARI.
2860 * Resource List Server support in the PJSIP stack, providing subscriptions to
2861 lists of resources and batched delivery of NOTIFY requests.
2862 * Inter-Asterisk distributed device state and mailbox state using the PJSIP
2865 It is important to note that Asterisk 13 is built on the architecture developed
2866 during the previous Standard release, Asterisk 12. Users upgrading to
2867 Asterisk 13 should read about the new features in Asterisk 12 later in this file
2868 (see Functionality changes from Asterisk 11 to Asterisk 12), as well as the
2869 UPGRADE-12.txt delivered with this release. In particular, users upgrading to
2870 Asterisk 13 from a release prior to Asterisk 12 should read the specifications
2871 on AMI, CDRs, and CEL on the Asterisk wiki:
2872 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
2873 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
2874 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
2876 Many new featuers in Asterisk 13 were introduced in point releases of
2877 Asterisk 12. Following this section - which documents the changes from all
2878 versions of Asterisk 12 to Asterisk 13 - users should examine the new features
2879 that were introduced in the point releases of Asterisk 12, as they are also
2880 included in Asterisk 13.
2882 Finally, all users upgrading to Asterisk 13 should read the UPGRADE.txt file
2883 delivered with this release.
2888 * Sample config files have been moved from configs/ to a sub-folder of that
2891 * The menuselect utility has been pulled into the Asterisk repository. As a
2892 result, the libxml2 development library is now a required dependency for
2895 * A new Compiler Flag, REF_DEBUG, has been added. When enabled, reference
2896 counted objects will emit additional debug information to the refs log file
2897 located in the standard Asterisk log file directory. This log file is useful
2898 in tracking down object leaks and other reference counting issues. Prior to
2899 this version, this option was only available by modifying the source code
2900 directly. This change also includes a new script, refcounter.py, in the
2901 contrib folder that will process the refs log file. Note that this replaces
2902 the refcounter utility that could be built from the utils directory.
2910 * This module was deprecated and has been removed. Users of app_dahdibarge
2911 should use ChanSpy instead.
2915 * New options to play a beep when starting a recording and stopping a recording
2916 have been added. The option "p" will play a beep to the channel that starts
2917 the recording. The option "P" will play a beep to the channel that stops the
2922 * Queue rules can now be stored in a database table, queue_rules. Unlike other
2923 RealTime tables, the queue_rules table is only examined on module load or
2924 module reload. A new general setting has been added to queuerules.conf,
2925 'realtime_rules', which, when set to 'yes', will cause app_queue to look in
2926 RealTime for additional queue rules to parse. Note that both the file and
2927 the database can be used as a provide of queue rules when 'realtime_rules'
2930 When app_queue is reloaded, all rules are re-parsed and loaded into memory.
2931 There is no caching of RealTime queue rules.
2935 * This module was deprecated and has been removed. Users of app_readfile
2936 should use func_env's FILE function instead.
2940 * The 'say' family of dialplan applications now support the Japanese
2941 language. The 'language' parameter in say.conf now recognizes a setting of
2942 'ja', which will enable Japanese language specific mechanisms for playing
2943 back numbers, dates, and other items.
2944 * Counting, enumeration and dates now supports Icelandic grammar with the
2945 'language' parameter set to 'is'.
2949 * This module was deprecated and has been removed. Users of app_saycountpl
2950 should use the Say family of applications.
2954 * The SetMusicOnHold dialplan application was deprecated and has been removed.
2955 Users of the application should use the CHANNEL function's musicclass
2960 * The WaitMusicOnHold dialplan application was deprecated and has been
2961 removed. Users of the application should use MusicOnHold with a duration
2966 * VoiceMail and VoiceMailMain now support the Japanese language. The
2967 'language' parameter in voicemail.conf now recognizes a setting of 'ja',
2968 which will enable prompts to be played back using a Japanese grammatical
2969 structure. Additional prompts are necessary for this functionality,
2971 - jb-arimasu: there is
2972 - jb-arimasen: there is not
2973 - jb-oshitekudasai: please press
2979 * Add the ability to specify multiple email addresses in configuration,
2988 * This module was deprecated and has been removed. Users of cdr_sqlite
2989 should use cdr_sqlite3_custom.
2993 * Added the ability to support PostgreSQL application_name on connections.
2994 This allows PostgreSQL to display the configured name in the
2995 pg_stat_activity view and CSV log entries. This setting is configurable
2996 for cdr_pgsql via the appname configuration setting in cdr_pgsql.conf.
3004 * Added the ability to support PostgreSQL application_name on connections.
3005 This allows PostgreSQL to display the configured name in the
3006 pg_stat_activity view and CSV log entries. This setting is configurable
3007 for cel_pgsql via the appname configuration setting in cel_pgsql.conf.
3015 * SS7 support now requires libss7 v2.0 or later.
3017 * Added SS7 support for connected line and redirecting.
3019 * Most SS7 CLI commands are reworked as well as new SS7 commands added.
3020 See online CLI help.
3022 * Added several SS7 config option parameters described in
3023 chan_dahdi.conf.sample.
3027 * This module was deprecated and has been removed. Users of chan_gtalk
3028 should use chan_motif.
3032 * This module was deprecated and has been removed. Users of chan_h323
3033 should use chan_ooh323.
3037 * This module was deprecated and has been removed. Users of chan_jingle
3038 should use chan_motif.
3042 * Added the CLI command 'pjsip list ciphers' so a user can know what
3043 OpenSSL names are available on their system for the pjsip.conf cipher
3048 * The SIPPEER dialplan function no longer supports using a colon as a
3049 delimiter for parameters. The parameters for the function should be
3050 delimited using a comma.
3052 * The SIPCHANINFO dialplan function was deprecated and has been removed. Users
3053 of the function should use the CHANNEL function instead.
3061 * Added functional peeraccount support. Except for Queue, the
3062 accountcode propagation is now consistently propagated to outgoing
3063 channels before dialing. The channel accountcode can change from its
3064 original non-empty value on channel creation for the following specific
3065 reasons. One, dialplan sets it using CHANNEL(accountcode). Two, an
3066 originate method that can specify an accountcode value. Three, the
3067 calling channel propagates its peeraccount or accountcode to the
3068 outgoing channel's accountcode before dialing. The change has two
3069 visible effects. One, local channels now cross accountcode and
3070 peeraccount across the special bridge between the ;1 and ;2 channels
3071 just like channels between normal bridges. Two, the
3072 CHANNEL(peeraccount) value can now be set before Dial and FollowMe to
3073 set the accountcode on the outgoing channel(s).
3075 For Queue, an outgoing channel's non-empty accountcode will not change
3076 unless explicitly set by CHANNEL(accountcode). The change has three
3077 visible effects. One, local channels now cross accountcode and
3078 peeraccount across the special bridge between the ;1 and ;2 channels
3079 just like channels between normal bridges. Two, the queue member will
3080 get an accountcode if it doesn't have one and one is available from the
3081 calling channel's peeraccount. Three, accountcode propagation includes
3082 local channel members where the accountcodes are propagated early
3083 enough to be available on the ;2 channel.
3087 * New DeviceStateChanged and PresenceStateChanged AMI events have been added.
3088 These events are emitted whenever a device state or presence state change
3089 occurs. The events are controlled by res_manager_device_state.so and
3090 res_manager_presence_state.so. If the high frequency of these events is
3091 problematic for you, do not load these modules.
3093 * Added DialplanExtensionAdd and DialplanExtensionRemove AMI commands. They
3094 work in basically the same way as the 'dialplan add extension' and
3095 'dialplan remove extension' CLI commands respectively.
3097 * New AMI action LoggerRotate reloads and rotates logger in the same manner
3098 as CLI command 'logger rotate'
3100 * New AMI Actions FAXSessions, FAXSession, and FAXStats replicate the
3101 functionality of CLI commands 'fax show sessions', 'fax show session',
3102 and fax show stats' respectively.
3104 * New AMI actions PRIDebugSet, PRIDebugFileSet, and PRIDebugFileUnset
3105 enable manager control over PRI debugging levels and file output.
3107 * AMI action PJSIPNotify may now send to a URI instead of only to a PJSIP
3108 endpoint as long as a default outbound endpoint is set. This also applies
3109 to the equivalent CLI command (pjsip send notify)
3111 * The AMI action PJSIPShowEndpoint now includes ContactStatusDetail sections
3112 that give information on Asterisk's attempts to qualify the endpoint.
3114 * The DialEnd event will now contain a Forward header if the dial is ending
3115 due to the call being forwarded. The contents of the Forward header is the
3116 extension in the number to which the call is being forwarded.
3120 * The "bridge_technology" extra field key has been added to BRIDGE_ENTER
3121 and BRIDGE_EXIT events.
3125 * Channel variables are now substituted in arguments passed to applications
3126 run by using dynamic features.
3130 * The TLS core in Asterisk now supports Perfect Forward Secrecy (PFS).
3131 Enabling PFS is attempted by default, and is dependent on the configuration
3132 of the module using TLS.
3133 - Ephemeral ECDH (ECDHE) is enabled by default. To disable it, do not
3134 specify a ECDHE cipher suite in sip.conf, for example:
3135 tlscipher=AES128-SHA:DES-CBC3-SHA
3136 - Ephemeral DH (DHE) is disabled by default. To enable it, add DH parameters
3137 into the private key file, e.g., sip.conf tlsprivatekey. For example, the
3138 default dh2048.pem - see
3139 http://www.opensource.apple.com/source/OpenSSL098/OpenSSL098-35.1/src/apps/dh2048.pem?txt
3140 - Because clients expect the server to prefer PFS, and because OpenSSL sorts
3141 its cipher suites by bit strength, see "openssl ciphers -v DEFAULT".
3142 Consider re-ordering your cipher suites in the respective configuration
3144 tlscipher=AES128+kEECDH:AES128+kEDH:3DES+kEDH:AES128-SHA:DES-CBC3-SHA:-ADH:-AECDH
3145 will use PFS when offered by the client. Clients which do not offer PFS
3146 fall-back to AES-128 (or even 3DES, as recommended by RFC 3261).
3154 * The JACK_HOOK function now supports audio with a sample rate higher than
3163 * Added the ability to support PostgreSQL application_name on connections.
3164 This allows PostgreSQL to display the configured name in the
3165 pg_stat_activity view and CSV log entries. This setting is configurable
3166 for res_config_pgsql via the dbappname configuration setting in
3169 res_pjsip_outbound_publish
3171 * A new module, res_pjsip_outbound_publish provides the mechanisms for sending
3172 PUBLISH requests for specific event packages to another SIP User Agent.
3176 * The publish/subscribe core module has been updated to support RFC 4662
3177 Resource Lists, allowing Asterisk to act as a Resource List Server (RLS).
3178 Resource lists are configured in pjsip.conf under a new object type,
3179 resource_list. Resource lists can contain either message-summary or presence
3180 events, and can be composed of specific resources that provide the event or
3181 other resource lists.
3183 * Inbound publication support is provided by a new object, inbound-publication.
3184 This configures res_pjsip_pubsub to accept PUBLISH requests from a particular
3185 resource. Which events are accepted is constructed dynamically; see
3186 res_pjsip_publish_asterisk for more information.
3188 res_pjsip_publish_asterisk
3190 * A new module, res_pjsip_publish_asterisk adds support for PUBLISH requests of
3191 Asterisk information to other Asterisk servers. This module is intended only
3192 for Asterisk to Asterisk exchanges of information. Currently, this includes
3193 both mailbox state and device state information.
3195 ------------------------------------------------------------------------------
3196 --- Functionality changes from Asterisk 12.4.0 to Asterisk 12.5.0 ------------
3197 ------------------------------------------------------------------------------
3201 * Stored recordings now support a new operation, copy. This will take an
3202 existing stored recording and copy it to a new location in the recordings
3205 * LiveRecording objects now have three additional fields that can be reported
3206 in a RecordingFinished ARI event:
3207 - total_duration: the duration of the recording
3208 - talking_duration: optional. The duration of talking detected in the
3209 recording. This is only available if max_silence_seconds was specified
3210 when the recording was started.
3211 - silence_duration: optional. The duration of silence detected in the
3212 recording. This is only available if max_silence_seconds was specified
3213 when the recording was started.
3214 Note that all duration values are reported in seconds.
3216 * Users of ARI can now send and receive out of call text messages. Messages
3217 can be sent directly to a particular endpoint, or can be sent to the
3218 endpoints resource directly and inferred from the URI scheme. Text
3219 messages are passed to ARI clients as TextMessageReceived events. ARI
3220 clients can choose to receive text messages by subscribing to the particular
3221 endpoint technology or endpoints that they are interested in.
3223 * The applications resource now supports subscriptions to all endpoints of
3224 a particular channel technology. For example, subscribing to an eventSource
3225 of 'endpoint:PJSIP' will subscribe to all PJSIP endpoints.
3229 * The endpoint configuration object now supports 'accountcode'. Any channel
3230 created for an endpoint with this setting will have its accountcode set
3231 to the specified value.
3235 * A new module, res_hep_rtcp, has been added that will forward RTCP call
3236 statistics to a HEP capture server. See res_hep for more information.
3240 * Function AUDIOHOOK_INHERIT has been deprecated. Audiohooks are now
3241 unconditionally inherited through masquerades. As a side benefit, more
3242 than one audiohook of a given type may persist through a masquerade now.
3244 ------------------------------------------------------------------------------
3245 --- Functionality changes from Asterisk 12.3.0 to Asterisk 12.4.0 ------------
3246 ------------------------------------------------------------------------------
3250 * Returns new AGENT_STATUS value "NOT_CONNECTED" if the agent fails to
3251 connect with an incoming caller after being alerted to the presence
3252 of the incoming caller. The most likely reason this would happen is
3253 the agent did not acknowledge the call in time.
3257 * New events have been added for the TALK_DETECT function. When the function
3258 is used on a channel, ChannelTalkingStart/ChannelTalkingStop events will be
3259 emitted to connected AMI clients indicating the start/stop of talking on
3264 * New event models have been aded for the TALK_DETECT function. When the
3265 function is used on a channel, ChannelTalkingStarted/ChannelTalkingFinished
3266 events will be emitted to connected WebSockets subscribed to the channel,
3267 indicating the start/stop of talking on the channel.
3271 * A new function, TALK_DETECT, has been added. When set on a channel, this
3272 fucntion causes events indicating the starting/stoping of talking on said
3273 channel to be emitted to both AMI and ARI clients.
3275 ------------------------------------------------------------------------------
3276 --- Functionality changes from Asterisk 12.2.0 to Asterisk 12.3.0 ------------
3277 ------------------------------------------------------------------------------
3281 * A new Playback URI 'tone' has been added. Tones are specified either as
3282 an indication name (e.g. 'tone:busy') from indications.conf or as a tone
3283 pattern (e.g. 'tone:240/250,0/250'). Tones differ from normal playback
3284 URIs in that they must be stopped manually and will continue to occupy
3285 a channel's ARI control queue until they are stopped. They also can not
3286 be rewound or fastforwarded.
3288 * User events can now be generated from ARI. Events can be signalled with
3289 arbitrary json variables, and include one or more of channel, bridge, or
3290 endpoint snapshots. An application must be specified which will receive
3291 the event message (other applications can subscribe to it). The message
3292 will also be delivered via AMI provided a channel is attached. Dialplan
3293 generated user event messages are still transmitted via the channel, and
3294 will only be received by a stasis application they are attached to or if
3295 the channel is subscribed to.
3299 * SIP peers can now specify 'trust_id_outbound' which affects RPID/PAI
3300 fields for prohibited callingpres information. Values are legacy, no, and
3301 yes. By default, legacy is used.
3302 trust_id_outbound=legacy - behavior remains the same as 1.8.26.1. When
3303 dealing with prohibited callingpres and sendrpid=pai/rpid, RPID/PAI
3304 headers are appended to outbound SIP messages just as they are with
3305 allowed callingpres values, but data about the remote party's identity is
3307 When sendrpid=rpid, only the remote party's domain is anonymized.
3308 trust_id_outbound=no - when dealing with prohibited callingpres, RPID/PAI
3309 headers are not sent.
3310 trust_id_outbound=yes - RPID/PAI headers are applied with the full remote
3311 party information in tact even for prohibited callingpres information.
3312 In the case of PAI, a Privacy: id header will be appended for prohibited
3313 calling information to communicate that the private information should
3314 not be relayed to untrusted parties.
3318 * Manager action 'Park' now takes an additional argument 'AnnounceChannel'
3319 which can be used to announce the parked call's location to an arbitrary
3320 channel in a bridge. If 'Channel' and 'TimeoutChannel' are now the two
3321 parties in a one to one bridge, 'TimeoutChannel' is treated as having
3322 parked 'Channel' like with the Park Call DTMF feature and will receive
3323 announcements prior to being hung up.
3325 ------------------------------------------------------------------------------
3326 --- Functionality changes from Asterisk 12.1.0 to Asterisk 12.2.0 ------------
3327 ------------------------------------------------------------------------------
3331 * Record application now has an option 'o' which allows 0 to act as an exit
3332 key setting the RECORD_STATUS variable to 'OPERATOR' instead of 'DTMF'
3335 --------------------------
3336 * ChanSpy now accepts a channel uniqueid or a fully specified channel name
3337 as the chanprefix parameter if the 'u' option is specified.
3340 --------------------------
3341 * CONFBRIDGE dialplan function is now capable of creating/modifying dynamic
3342 conference user menus.
3344 * CONFBRIDGE dialplan function is now capable of removing dynamic conference
3345 menus, bridge settings, and user settings that have been applied by the
3346 CONFBRIDGE dialplan function.
3348 * The ConfBridge dialplan application now sets a channel variable,
3349 CONFBRIDGE_RESULT, upon exiting. This variable can be used to determine
3350 how a channel exited the conference.
3352 * Added conference user option 'announce_join_leave_review'. This option
3353 implies 'announce_join_leave' with the added effect that the user will
3354 be asked if they want to confirm or re-record the recording of their
3355 name when entering the conference
3358 --------------------------
3359 * At exit, the Directory application now sets a channel variable
3360 DIRECTORY_RESULT to one of the following based on the reason for exiting:
3361 OPERATOR user requested operator by pressing '0' for operator
3362 ASSISTANT user requested assistant by pressing '*' for assistant
3363 TIMEOUT user pressed nothing and Directory stopped waiting
3364 HANGUP user's channel hung up
3365 SELECTED user selected a user from the directory and is routed
3366 USEREXIT user pressed '#' from the selection prompt to exit
3367 FAILED directory failed in a way that wasn't accounted for. Dang.
3371 * Monitor() - A new option, B(), has been added that will turn on a periodic
3372 beep while the call is being recorded.
3375 --------------------------
3376 * MusicOnHold streams (all modes other than "files") now support wide band
3380 --------------------------
3381 * Added options 'b' and 'B' to apply predial handlers for outgoing calls
3382 and for the channel executing Page respectively.
3385 --------------------------
3386 * PickupChan now accepts channel uniqueids of channels to pickup.
3389 --------------------------
3390 * If a channel variable SAY_DTMF_INTERRUPT is present on a channel and set
3391 to 'true' (case insensitive), then any Say application (SayNumber,
3392 SayDigits, SayAlpha, SayAlphaCase, SayUnixTime, and SayCounted) will
3393 anticipate DTMF. If DTMF is received, these applications will behave like
3394 the background application and jump to the received extension once a match
3395 is established or after a short period of inactivity.
3398 -------------------------
3399 * A new function, MIXMONITOR, has been added to allow access to individual
3400 instances of MixMonitor on a channel.
3402 * A new option, B(), has been added that will turn on a periodic beep while the
3403 call is being recorded.
3407 -------------------------
3410 -------------------------
3411 * TEL URI support for inbound INVITE requests has been added. chan_sip will
3412 now handle TEL schemes in the Request and From URIs. The phone-context in
3413 the Request URI will be stored in the SIPURIPHONECONTEXT channel variable on
3414 the inbound channel.
3418 * Exposed sorcery-based configuration files like pjsip.conf to dialplans via
3419 the new AST_SORCERY diaplan function.
3421 * Core Show Locks output now includes Thread/LWP ID if the platform
3422 supports this feature.
3424 * New "logger add channel" and "logger remove channel" CLI commands have
3425 been added to allow creation and deletion of dynamic logger channels
3426 without configuration changes. These dynamic logger channels will only
3427 exist until the next restart of asterisk.
3431 * The live recording object on recording events now contains a target_uri
3432 field which contains the URI of what is being recorded.
3434 * The bridge type used when creating a bridge is now a comma separated list of
3435 bridge properties. Valid options are: mixing, holding, dtmf_events, and
3438 * A channelId can now be provided when creating a channel, either in the
3439 uri (POST channels/my-channel-id) or as query parameter. A local channel
3440 will suffix the second channel id with ';2' unless provided as query
3441 parameter otherChannelId.
3443 * A bridgeId can now be provided when creating a bridge, either in the uri
3444 (POST bridges/my-bridge-id) or as a query parameter.
3446 * A playbackId can be provided when starting a playback, either in the uri
3447 (POST channels/my-channel-id/play/my-playback-id /
3448 POST bridges/my-bridge-id/play/my-playback-id) or as a query parameter.
3450 * A snoop channel can be started with a snoopId, in the uri or query.
3454 * Originate now takes optional parameters ChannelId and OtherChannelId,
3455 used to set the UniqueId on creation. The other id is assigned to the
3456 second channel when dialing LOCAL, or defaults to appending ;2 if only
3457 the single Id is given.
3459 * The Mixmonitor action now has a "Command" header that can be used to
3460 indicate a post-process command to run once recording finishes.
3464 * A new set of Alembic scripts has been added for CDR tables. This will create
3465 a 'cdr' table with the default schema that Asterisk expects.
3470 * A new function was added: PERIODIC_HOOK. This allows running a periodic
3471 dialplan hook on a channel. Any audio generated by this hook will be
3472 injected into the call.
3480 * A new module, res_hep, has been added, that acts as a generic packet
3481 capture agent for the Homer Encapsulation Protocol (HEP) version 3.
3482 It can be configured via hep.conf. Other modules can use res_hep to send
3483 message traffic to a HEP capture server.
3487 * A new module, res_hep_pjsip, has been added that will forward PJSIP
3488 message traffic to a HEP capture server. See res_hep for more
3493 * transport and endpoint ToS options (tos, tos_audio, and tos_video) may now
3494 be set as the named set of ToS values (cs0-cs7, af11-af43, ef).
3496 * Added the following new CLI commands:
3497 - "pjsip show contacts" - list all current PJSIP contacts.
3498 - "pjsip show contact" - show specific information about a current PJSIP
3500 - "pjsip show channel" - show detailed information about a PJSIP channel.
3502 res_pjsip_multihomed
3504 * A new module, res_pjsip_multihomed handles situations where the system
3505 Asterisk is running out has multiple interfaces. res_pjsip_multihomed
3506 determines which interface should be used during message sending.
3508 res_pjsip_pidf_digium_body_supplement
3510 * A new module, res_pjsip_pidf_digium_body_supplement provides NOTIFY
3511 request body formatting for presence support in Digium phones.
3513 res_pjsip_send_to_voicemail
3515 * A new module, res_pjsip_send_to_voicemail allows for REFER requests with
3516 particular headers to transfer a PJSIP channel directly to a particular
3517 extension that has VoiceMail. This is intended to be used with Digium
3518 phones that support this feature.
3520 res_pjsip_outbound_registration
3522 * A new CLI command has been added: "pjsip show registrations", which lists
3523 all configured PJSIP registrations
3526 ------------------------------------------------------------------------------
3527 --- Functionality changes from Asterisk 12.0.0 to Asterisk 12.1.0 ------------
3528 ------------------------------------------------------------------------------
3532 * Added a new module that provides AMI control over MWI within Asterisk,
3533 res_mwi_external_ami. Note that this module depends on res_mwi_external;
3534 for more information on enabling this module, see res_mwi_external.
3535 This module provides the MWIGet/MWIUpdate/MWIDelete actions, as well as
3536 the MWIGet/MWIGetComplete events.
3538 * The DialStatus field in the DialEnd event can now contain additional
3539 statuses that convey how the dial operation terminated. This includes
3540 ABORT, CONTINUE, and GOTO.
3542 * AMI will now emit security events. A new class authorization has been
3543 added in manager.conf for the security events, 'security'. The new events
3545 - FailedACL - raised when a request violates an ACL check
3546 - InvalidAccountID - raised when a request fails an authentication
3547 check due to an invalid account ID
3548 - SessionLimit - raised when a request fails due to exceeding the
3549 number of allowed concurrent sessions for a service
3550 - MemoryLimit - raised when a request fails due to an internal memory
3552 - LoadAverageLimit - raised when a request fails because a configured
3553 load average limit has been reached
3554 - RequestNotAllowed - raised when a request is not allowed by
3556 - AuthMethodNotAllowed - raised when a request used an authentication
3557 method not allowed by the service
3558 - RequestBadFormat - raised when a request is received with bad formatting
3559 - SuccessfulAuth - raised when a request successfully authenticates
3560 - UnexpectedAddress - raised when a request has a different source address
3561 then what is expected for a session already in progress with a service
3562 - ChallengeResponseFailed - raised when a request's attempt to authenticate
3563 has been challenged, and the request failed the authentication challenge
3564 - InvalidPassword - raised when a request provides an invalid password
3565 during an authentication attempt
3566 - ChallengeSent - raised when an Asterisk service send an authentication
3567 challenge to a request
3568 - InvalidTransport - raised when a request attempts to use a transport not
3569 allowed by the Asterisk service
3571 * Bridge related events now have two additional fields: BridgeName and
3572 BridgeCreator. BridgeName is a descriptive name for the bridge;
3573 BridgeCreator is the name of the entity that created the bridge. This
3574 affects the following events: ConfbridgeStart, ConfbridgeEnd,
3575 ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord,
3576 ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer,
3577 AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave
3581 * The Bridge data model now contains the additional fields 'name' and
3582 'creator'. The 'name' field conveys a descriptive name for the bridge;
3583 the 'creator' field conveys the name of the entity that created the bridge.
3584 This affects all responses to HTTP requests that return a Bridge data model
3585 as well as all event derived data models that contain a Bridge data model.
3586 The POST /bridges operation may now optionally specify a name to give to
3587 the bridge being created.
3589 * Added a new ARI resource 'mailboxes' which allows the creation and
3590 modification of mailboxes managed by external MWI. Modules res_mwi_external
3591 and res_stasis_mailbox must be enabled to use this resource. For more
3592 information on external MWI control, see res_mwi_external.
3594 * Added new events for externally initiated transfers. The event
3595 BridgeBlindTransfer is now raised when a channel initiates a blind transfer
3596 of a bridge in the ARI controlled application to the dialplan; the
3597 BridgeAttendedTransfer event is raised when a channel initiates an
3598 attended transfer of a bridge in the ARI controlled application to the
3601 * Channel variables may now be specified as a body parameter to the
3602 POST /channels operation. The 'variables' key in the JSON is interpreted
3603 as a sequence of key/value pairs that will be added to the created channel
3604 as channel variables. Other parameters in the JSON body are treated as
3605 query parameters of the same name.
3609 * Asterisk's HTTP server now supports chunked Transfer-Encoding. This will be
3610 automatically handled by the HTTP server if a request is received with a
3611 Transfer-Encoding type of "chunked".
3615 * Path support has been added with the 'support_path' option in registration
3618 * A 'debug' option has been added to the globals section that will allow
3619 sip messages to be logged.
3621 * A 'set_var' option has been added to endpoints that will automatically
3622 set the desired variable(s) on a channel created for that endpoint.
3624 * Several new tables and columns have been added to the realtime schema for
3625 the res_pjsip related modules. See the UPGRADE.txt notes for updating
3626 the database schema.
3630 * A new module, res_mwi_external, has been added to Asterisk. This module
3631 acts as a base framework that other modules can build on top of to allow
3632 an external system to control MWI within Asterisk. For implementations
3633 that make use of res_mwi_external, see res_mwi_external_ami and
3634 res_ari_mailboxes. Note that res_mwi_external conflicts with other modules
3635 that may produce MWI themselves, such as app_voicemail. res_mwi_external
3636 and other modules that depend on it cannot be built or loaded with
3637 app_voicemail present.
3641 * DNS functionality will now automatically be enabled if the system configured
3642 nameservers can be retrieved. If the system configured nameservers can not be
3643 retrieved the functionality will resort to using system resolution. Functionality
3644 such as SRV records and failover will not be available if system resolution
3647 ------------------------------------------------------------------------------
3648 --- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
3649 ------------------------------------------------------------------------------
3654 Asterisk 12 is a standard release of the Asterisk project. As such, the
3655 focus of development for this release was on core architectural changes and
3656 major new features. This includes:
3657 * A more flexible bridging core based on the Bridging API
3658 * A new internal message bus, Stasis
3659 * Major standardization and consistency improvements to AMI
3660 * Addition of the Asterisk RESTful Interface (ARI)
3661 * A new SIP channel driver, chan_pjsip
3662 In addition, as the vast majority of bridging in Asterisk was migrated to the
3663 Bridging API used by ConfBridge, major changes were made to most of the
3664 interfaces in Asterisk. This includes not only AMI, but also CDRs and CEL.
3666 Specifications have been written for the affected interfaces. These
3667 specifications are available on the Asterisk wiki:
3668 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
3669 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
3670 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
3672 It is *highly* recommended that anyone migrating to Asterisk 12 read the
3673 information regarding its release both in this file and in the accompanying
3674 UPGRADE.txt file. More detailed information on the major changes can be found
3675 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/0YCLAQ.
3680 * Added build option DISABLE_INLINE. This option can be used to work around a
3681 bug in gcc. For more information, see
3682 http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816
3684 * Removed the CHANNEL_TRACE development mode build option. Certain aspects of
3685 the CHANNEL_TRACE build option were incompatible with the new bridging
3688 * Asterisk now optionally uses libxslt to improve XML documentation generation
3689 and maintainability. If libxslt is not available on the system, some XML
3690 documentation will be incomplete.
3692 * Asterisk now depends on libjansson. If a package of libjansson is not
3693 available on your distro, please see http://www.digip.org/jansson/.
3695 * Asterisk now depends on libuuid and, optionally, uriparser. It is
3696 recommended that you install uriparser, even if it is optional.
3698 * The new SIP stack and channel driver uses a particular version of PJSIP.
3699 Please see https://wiki.asterisk.org/wiki/x/J4GLAQ for more information on
3700 configuring and installing PJSIP for usage with Asterisk.
3702 * Optional API was re-implemented to be more portable, and no longer requires
3703 weak reference support from the compiler. The build option OPTIONAL_API may
3704 be disabled to disable Optional API support.
3711 * Along with AgentRequest, this application has been modified to be a
3712 replacement for chan_agent. The act of a channel calling the AgentLogin
3713 application places the channel into a pool of agents that can be
3714 requested by the AgentRequest application. Note that this application, as
3715 well as all other agent related functionality, is now provided by the
3716 app_agent_pool module. See chan_agent and AgentRequest for more information.
3718 * This application no longer performs agent authentication. If authentication
3719 is desired, the dialplan needs to perform this function using the
3720 Authenticate or VMAuthenticate application or through an AGI script before
3723 * If this application is called and the agent is already logged in, the
3724 dialplan will continue execution with the AGENT_STATUS channel variable set
3725 to ALREADY_LOGGED_IN.
3727 * The agents.conf schema has changed. Rather than specifying agents on a
3728 single line in comma delineated fashion, each agent is defined in a separate
3729 context. This allows agents to use the power of context templates in their
3732 * A number of parameters from agents.conf have been removed. This includes
3733 maxloginretries, autologoffunavail, updatecdr, goodbye, group, recordformat,
3734 urlprefix, and savecallsin. These options were obsoleted by the move from
3735 a channel driver model to the bridging/application model provided by
3740 * A new application, this will request a logged in agent from the pool and
3741 bridge the requested channel with the channel calling this application.
3742 Logged in agents are those channels that called the AgentLogin application.
3743 If an agent cannot be requested from the pool, the AGENT_STATUS dialplan
3744 application will be set with an appropriate error value.
3746 AgentMonitorOutgoing
3748 * This application has been removed. It was a holdover from when
3749 AgentCallbackLogin was removed.
3753 * Added support for additional Ademco DTMF signalling formats, including
3754 Express 4+1, Express 4+2, High Speed and Super Fast.
3756 * Added channel variable ALARMRECEIVER_CALL_LIMIT. This sets the maximum
3757 call time, in milliseconds, to run the application.
3759 * Added channel variable ALARMRECEIVER_RETRIES_LIMIT. This sets the
3760 maximum number of times to retry the call.
3762 * Added a new configuration option answait. If set, the AlarmReceiver
3763 application will wait the number of milliseconds specified by answait
3764 after the channel has answered. Valid values range between 500
3765 milliseconds and 10000 milliseconds.
3767 * Added configuration option no_group_meta. If enabled, grouping of metadata
3768 information in the AlarmReceiver log file will be skipped.
3772 * It is now no longer possible to bypass updating the CDR on the channel
3773 when answering. CDRs reflect the state of the channel and will always
3774 reflect the time they were Answered.
3778 * A new application in Asterisk, this will place the calling channel
3779 into a holding bridge, optionally entertaining them with some form of
3780 media. Channels participating in a holding bridge do not interact with
3781 other channels in the same holding bridge. Optionally, however, a channel
3782 may join as an announcer. Any media passed from an announcer channel is
3783 played to all channels in the holding bridge. Channels leave a holding
3784 bridge either when an optional timer expires, or via the ChannelRedirect
3785 application or AMI Redirect action.
3789 * All participants in a bridge can now be kicked out of a conference room
3790 by specifying the channel parameter as 'all' in the ConfBridge kick CLI
3791 command, i.e., 'confbridge kick <conference> all'
3793 * CLI output for the 'confbridge list' command has been improved. When
3794 displaying information about a particular bridge, flags will now be shown
3795 for the participating users indicating properties of that user.
3797 * The ConfbridgeList event now contains the following fields: WaitMarked,
3798 EndMarked, and Waiting. This displays additional properties about the
3799 user's profile, as well as whether or not the user is waiting for a
3800 Marked user to enter the conference.
3802 * Added a new option for conference recording, record_file_append. If enabled,
3803 when the recording is stopped and then re-started, the existing recording
3804 will be used and appended to.
3806 * ConfBridge now has the ability to set the language of announcements to the
3807 conference. The language can be set on a bridge profile in confbridge.conf
3808 or by the dialplan function CONFBRIDGE(bridge,language)=en.
3812 * The channel variable CPLAYBACKSTATUS may now return the value
3813 'REMOTESTOPPED'. This occurs when playback is stopped by a remote interface,
3814 such as AMI. See the AMI action ControlPlayback for more information.
3818 * Added the 'a' option, which allows the caller to enter in an additional
3819 alias for the user in the directory. This option must be used in conjunction
3820 with the 'f', 'l', or 'b' options. Note that the alias for a user can be
3821 specified in voicemail.conf.
3825 * The output of DumpChan no longer includes the DirectBridge or IndirectBridge
3826 fields. Instead, if a channel is in a bridge, it includes a BridgeID field
3827 containing the unique ID of the bridge that the channel happens to be in.
3831 * ForkCDR no longer automatically resets the forked CDR. See the 'r' option
3832 for more information.
3834 * Variables are no longer purged from the original CDR. See the 'v' option for
3837 * The 'A' option has been removed. The Answer time on a CDR is never updated
3840 * The 'd' option has been removed. The disposition on a CDR is a function of
3841 the state of the channel and cannot be altered.
3843 * The 'D' option has been removed. Who the Party B is on a CDR is a function
3844 of the state of the respective channels involved in the CDR and cannot be
3847 * The 'r' option has been changed. Previously, ForkCDR always reset the CDR
3848 such that the start time and, if applicable, the answer time was updated.
3849 Now, by default, ForkCDR simply forks the CDR, maintaining any times. The
3850 'r' option now triggers the Reset, setting the start time (and answer time
3851 if applicable) to the current time. Note that the 'a' option still sets
3852 the answer time to the current time if the channel was already answered.
3854 * The 's' option has been removed. A variable can be set on the original CDR
3855 if desired using the CDR function, and removed from a forked CDR using the
3858 * The 'T' option has been removed. The concept of DONT_TOUCH and LOCKED no
3859 longer applies in the CDR engine.
3861 * The 'v' option now prevents the copy of the variables from the original CDR
3862 to the forked CDR. Previously the variables were always copied but were
3863 removed from the original. This was changed as removing variables from a CDR
3864 can have unintended side effects - this option allows the user to prevent
3865 propagation of variables from the original to the forked without modifying
3870 * Added the 'n' option to MeetMe to prevent application of the DENOISE
3871 function to a channel joining a conference. Some channel drivers that vary
3872 the number of audio samples in a voice frame will experience significant
3873 quality problems if a denoiser is attached to the channel; this option gives
3874 them the ability to remove the denoiser without having to unload func_speex.
3878 * The 'b' option now includes conferences as well as sounds played to the
3881 * The AUDIOHOOK_INHERIT function is no longer needed to keep a MixMonitor
3882 running during a transfer. If a MixMonitor is started on a channel,
3883 the MixMonitor will continue to record the audio passing through the
3884 channel even in the presence of transfers.
3888 * The NoCDR application is deprecated. Please use the CDR_PROP function to
3891 * While the NoCDR application will prevent CDRs for a channel from being
3892 propagated to registered CDR backends, it will not prevent that data from
3893 being collected. Hence, a subsequent call to ResetCDR or the CDR_PROP
3894 function that enables CDRs on a channel will restore those records that have
3895 not yet been finalized.
3899 * The app_parkandannounce module has been removed. The application
3900 ParkAndAnnounce is now provided by the res_parking module. See the
3901 res_parking changes for more information.
3905 * Added queue available hint. The hint can be added to the dialplan using the
3906 following syntax: exten,hint,Queue:{queue_name}_avail
3907 For example, if the name of the queue is 'markq':
3908 exten => 8501,hint,Queue:markq_avail
3909 This will report 'InUse' if there are no logged in agents or no free agents.
3910 It will report 'Idle' when an agent is free.
3912 * Queues now support a hint for member paused state. The hint uses the form
3913 'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name}
3914 are the name of the queue and the name of the member to subscribe to,
3915 respectively. For example: exten => 8501,hint,Queue:sales_pause_mark.
3916 Members will show as In Use when paused.
3918 * The configuration options eventwhencalled and eventmemberstatus have been
3919 removed. As a result, the AMI events QueueMemberStatus, AgentCalled,
3920 AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be
3921 sent. The "Variable" fields will also no longer exist on the Agent* events.
3922 These events can be filtered out from a connected AMI client using the
3923 eventfilter setting in manager.conf.
3925 * The queue log now differentiates between blind and attended transfers. A
3926 blind transfer will result in a BLINDTRANSFER message with the destination
3927 context and extension. An attended transfer will result in an
3928 ATTENDEDTRANSFER message. This message will indicate the method by which
3929 the attended transfer was completed: "BRIDGE" for a bridge merge, "APP"
3930 for running an application on a bridge or channel, or "LINK" for linking
3931 two bridges together with local channels. The queue log will also now detect
3932 externally initiated blind and attended transfers and record the transfer
3935 * When performing queue pause/unpause on an interface without specifying an
3936 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
3937 least one member of any queue exists for that interface.
3939 * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
3940 for realtime queue log entries.
3944 * The 'e' option has been deprecated. Use the CDR_PROP function to re-enable
3945 CDRs when they were previously disabled on a channel.
3947 * The 'w' and 'a' options have been removed. Dispatching CDRs to registered
3948 backends occurs on an as-needed basis in order to preserve linkedid
3949 propagation and other needed behavior.
3953 * A new application, this is similar to SayAlpha except that it supports
3954 case sensitive playback of the specified characters. For example,
3955 SayAlphaCase(u,aBc) will result in 'a uppercase b c'.
3959 * This application is deprecated in favor of CHANNEL(amaflags).
3963 * The SendDTMF application will now accept 'W' as valid input. This will cause
3964 the application to delay one second while streaming DTMF.
3968 * A new application in Asterisk 12, this hands control of the channel calling
3969 the application over to an external system. Currently, external systems
3970 manipulate channels in Stasis through the Asterisk RESTful Interface (ARI).
3974 * UserEvent will now handle duplicate keys by overwriting the previous value
3975 assigned to the key.
3977 * In addition to AMI, UserEvent invocations will now be distributed to any
3978 interested Stasis applications.
3982 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
3983 system as mailbox@context. The rest of the system cannot add @default
3984 to mailbox identifiers for app_voicemail that do not specify a context
3985 any longer. It is a mailbox identifier format that should only be
3986 interpreted by app_voicemail.
3988 * The voicemail.conf configuration file now has an 'alias' configuration
3989 parameter for use with the Directory application. The voicemail realtime
3990 database table schema has also been updated with an 'alias' column.
3995 * Pass through support has been added for both VP8 and Opus.
3997 * Added format attribute negotiation for the Opus codec. Format attribute
3998 negotiation is provided by the res_format_attr_opus module.
4003 * Masquerades as an operation inside Asterisk have been effectively hidden
4004 by the migration to the Bridging API. As such, many 'quirks' of Asterisk
4005 no longer occur. This includes renaming of channels, "<ZOMBIE>" channels,
4006 dropping of frame/audio hooks, and other internal implementation details
4007 that users had to deal with. This fundamental change has large implications
4008 throughout the changes documented for this version. For more information
4009 about the new core architecture of Asterisk, please see the Asterisk wiki.
4011 * Multiple parties in a bridge may now be transferred. If a participant in a
4012 multi-party bridge initiates a blind transfer, a Local channel will be used
4013 to execute the dialplan location that the transferer sent the parties to. If
4014 a participant in a multi-party bridge initiates an attended transfer,
4015 several options are possible. If the attended transfer results in a transfer
4016 to an application, a Local channel is used. If the attended transfer results
4017 in a transfer to another channel, the resulting channels will be merged into
4020 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel
4021 driver specific. If the channel variable is set on the transferrer channel,
4022 the sound will be played to the target of an attended transfer.
4024 * The channel variable BRIDGEPEER becomes a comma separated list of peers in
4025 a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers
4026 listed. Any more peers in the bridge will not be included in the list.
4027 BRIDGEPEER is not valid in holding bridges like parking since those channels
4028 do not talk to each other even though they are in a bridge.
4030 * The channel variable BRIDGEPVTCALLID is only valid for two party bridges
4031 and will contain a value if the BRIDGEPEER's channel driver supports it.
4033 * A channel variable ATTENDEDTRANSFER is now set which indicates which channel
4034 was responsible for an attended transfer in a similar fashion to
4037 * Modules using the Configuration Framework or Sorcery must have XML
4038 configuration documentation. This configuration documentation is included
4039 with the rest of Asterisk's XML documentation, and is accessible via CLI
4040 commands. See the CLI changes for more information.
4042 AMI (Asterisk Manager Interface)
4044 * Major changes were made to both the syntax as well as the semantics of the
4045 AMI protocol. In particular, AMI events have been substantially improved
4046 in this version of Asterisk. For more information, please see the AMI
4047 specification at https://wiki.asterisk.org/wiki/x/dAFRAQ
4049 * AMI events that reference a particular channel or bridge will now always
4050 contain a standard set of fields. When multiple channels or bridges are
4051 referenced in an event, fields for at least some subset of the channels
4052 and bridges in the event will be prefixed with a descriptive name to avoid
4053 name collisions. See the AMI event documentation on the Asterisk wiki for
4056 * The CLI command 'manager show commands' no longer truncates command names
4057 longer than 15 characters and no longer shows authorization requirement
4058 for commands. 'manager show command' now displays the privileges needed
4059 for using a given manager command instead.
4061 * The SIPshowpeer action will now include a 'SubscribeContext' field for a
4062 peer in its response if the peer has a subscribe context set.
4064 * The SIPqualifypeer action now acknowledges the request once it has
4065 established that the request is against a known peer. It also issues a new
4066 event, 'SIPQualifyPeerDone', once the qualify action has been completed.
4068 * The PlayDTMF action now supports an optional 'Duration' parameter. This
4069 specifies the duration of the digit to be played, in milliseconds.
4071 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
4072 updates when changes occur instead of requiring the use of pollmailboxes.
4074 * Added a new action 'ControlPlayback'. The ControlPlayback action allows an
4075 AMI client to manipulate audio currently being played back on a channel. The
4076 supported operations depend on the application being used to send audio to
4077 the channel. When the audio playback was initiated using the ControlPlayback
4078 application or CONTROL STREAM FILE AGI command, the audio can be paused,
4079 stopped, restarted, reversed, or skipped forward. When initiated by other
4080 mechanisms (such as the Playback application), the audio can be stopped,
4081 reversed, or skipped forward.
4083 * Channel related events now contain a snapshot of channel state, adding new
4084 fields to many of these events.
4086 * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
4087 in a future release. Please use the common 'Exten' field instead.
4089 * The AMI event 'UserEvent' from app_userevent now contains the channel state
4090 fields. The channel state fields will come before the body fields.
4092 * The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
4093 'UnParkedCall' have changed significantly in the new res_parking module.
4095 The 'Channel' and 'From' headers are gone. For the channel that was parked
4096 or is coming out of parking, a 'Parkee' channel snapshot is issued and it
4097 has a number of fields associated with it. The old 'Channel' header relayed
4098 the same data as the new 'ParkeeChannel' header.
4100 The 'From' field was ambiguous and changed meaning depending on the event.
4101 for most of these, it was the name of the channel that parked the call
4102 (the 'Parker'). There is no longer a header that provides this channel name,
4103 however the 'ParkerDialString' will contain a dialstring to redial the
4104 device that parked the call.
4106 On UnParkedCall events, the 'From' header would instead represent the
4107 channel responsible for retrieving the parkee. It receives a channel
4108 snapshot labeled 'Retriever'. The 'from' field is is replaced with
4111 Lastly, the 'Exten' field has been replaced with 'ParkingSpace'.
4113 * The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar
4114 fashion has changed the field names 'StartExten' and 'StopExten' to
4115 'StartSpace' and 'StopSpace' respectively.
4117 * The deprecated use of | (pipe) as a separator in the channelvars setting in
4118 manager.conf has been removed.
4120 * Channel Variables conveyed with a channel no longer contain the name of the
4121 channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now
4122 ChanVariable: bar=baz. When multiple channels are present in a single AMI
4123 event, the various ChanVariable fields will contain a suffix that specifies
4124 which channel they correspond to.
4126 * The NewPeerAccount AMI event is no longer raised. The NewAccountCode AMI
4127 event always conveys the AMI event for a particular channel.
4129 * All 'Reload' events have been consolidated into a single event type. This
4130 event will always contain a Module field specifying the name of the module
4131 and a Status field denoting the result of the reload. All modules now issue
4132 this event when being reloaded.
4134 * The 'ModuleLoadReport' event has been removed. Most AMI connections would
4135 fail to receive this event due to being connected after modules have loaded.
4136 AMI connections that want to know when Asterisk is ready should listen for
4137 the 'FullyBooted' event.
4139 * app_fax now sends the same send fax/receive fax events as res_fax. The
4140 'FaxSent' event is now the 'SendFAX' event, and the 'FaxReceived' event is
4141 now the 'ReceiveFAX' event.
4143 * The 'MusicOnHold' event is now two events: 'MusicOnHoldStart' and
4144 'MusicOnHoldStop'. The sub type field has been removed.
4146 * The 'JabberEvent' event has been removed. It is not AMI's purpose to be a
4147 carrier for another protocol.
4149 * The Bridge Manager action's 'Playtone' header now accepts more fine-grained
4150 options. 'Channel1' and 'Channel2' may be specified in order to play a tone
4151 to the specific channel. 'Both' may be specified to play a tone to both
4152 channels. The old 'yes' option is still accepted as a way of playing the
4153 tone to Channel2 only.
4155 * The AMI 'Status' response event to the AMI Status action replaces the
4156 'BridgedChannel' and 'BridgedUniqueid' headers with the 'BridgeID' header to
4157 indicate what bridge the channel is currently in.
4159 * The AMI 'Hold' event has been moved out of individual channel drivers, into
4160 core, and is now two events: 'Hold' and 'Unhold'. The status field has been
4163 * The AMI events in app_queue have been made more consistent with each other.
4164 Events that reference channels (QueueCaller* and Agent*) will show
4165 information about each channel. The (infamous) 'Join' and 'Leave' AMI
4166 events have been changed to 'QueueCallerJoin' and 'QueueCallerLeave'.
4168 * The 'MCID' AMI event now publishes a channel snapshot when available and
4169 its non-channel-snapshot parameters now use either the "MCallerID" or
4170 'MConnectedID' prefixes with Subaddr*, Name*, and Num* suffixes instead
4171 of 'CallerID' and 'ConnectedID' to avoid confusion with similarly named
4172 parameters in the channel snapshot.
4174 * The AMI events 'Agentlogin' and 'Agentlogoff' have been renamed
4175 'AgentLogin' and 'AgentLogoff' respectively.
4177 * The 'Channel' key used in the 'AlarmClear', 'Alarm', and 'DNDState' has been
4178 renamed "DAHDIChannel" since it does not convey an Asterisk channel name.
4180 * 'ChannelUpdate' events have been removed.
4182 * All AMI events now contain a 'SystemName' field, if available.
4184 * Local channel optimization is now conveyed in two events:
4185 'LocalOptimizationBegin' and 'LocalOptimizationEnd'. The Begin event is sent
4186 when the Local channel driver begins attempting to optimize itself out of
4187 the media path; the End event is sent after the channel halves have
4188 successfully optimized themselves out of the media path.
4190 * Local channel information in events is now prefixed with 'LocalOne' and
4191 'LocalTwo'. This replaces the suffix of '1' and '2' for the two halves of
4192 the Local channel. This affects the 'LocalBridge', 'LocalOptimizationBegin',
4193 and 'LocalOptimizationEnd' events.
4195 * The option 'allowmultiplelogin' can now be set or overriden in a particular
4196 account. When set in the general context, it will act as the default
4197 setting for defined accounts.
4199 * The 'BridgeAction' event was removed. It technically added no value, as the
4200 Bridge Action already receives confirmation of the bridge through a
4201 successful completion Event.
4203 * The 'BridgeExec' events were removed. These events duplicated the events that
4204 occur in the Bridging API, and are conveyed now through BridgeCreate,
4205 BridgeEnter, and BridgeLeave events.
4207 * The 'RTCPSent'/'RTCPReceived' events have been significantly modified from
4208 previous versions. They now report all SR/RR packets sent/received, and
4209 have been restructured to better reflect the data sent in a SR/RR. In
4210 particular, the event structure now supports multiple report blocks.
4212 * Added 'BlindTransfer' and 'AttendedTransfer' events. These events are
4213 raised when a blind transfer/attended transfer completes successfully.
4214 They contain information about the transfer that just completed, including
4215 the location of the transfered channel.
4217 * Added a 'security' class to AMI which outputs the required fields for
4218 security messages similar to the log messages from res_security_log
4220 * The AMI event 'ExtensionStatus' now contains a 'StatusText' field
4221 that describes the status value in a human readable string.
4223 CDR (Call Detail Records)
4225 * Significant changes have been made to the behavior of CDRs. The CDR engine
4226 was effectively rewritten and built on the Stasis message bus. For a full
4227 definition of CDR behavior in Asterisk 12, please read the specification
4228 on the Asterisk wiki (wiki.asterisk.org).
4230 * CDRs will now be created between all participants in a bridge. For each
4231 pair of channels in a bridge, a CDR is created to represent the path of
4232 communication between those two endpoints. This lets an end user choose who
4233 to bill for what during bridge operations with multiple parties.
4235 * The duration, billsec, start, answer, and end times now reflect the times
4236 associated with the current CDR for the channel, as opposed to a cumulative
4237 measurement of all CDRs for that channel.
4239 * When a CDR is dispatched, user defined CDR variables from both parties are
4240 included in the resulting CDR. If both parties have the same variable, only
4241 the Party A value is provided.
4243 * Added a new option to cdr.conf, 'debug'. When enabled, significantly more
4244 information regarding the CDR engine is logged as verbose messages. This
4245 option should only be used if the behavior of the CDR engine needs to be
4248 * Added CLI command 'cdr set debug {on|off}'. This toggles the 'debug' setting
4249 normally configured in cdr.conf.
4251 * Added CLI command 'cdr show active {channel}'. When {channel} is not
4252 specified, this command provides a summary of the channels with CDR
4253 information and their statistics. When {channel} is specified, it shows
4254 detailed information about all records associated with {channel}.
4256 CEL (Channel Event Logging)
4258 * CEL has undergone significant rework in Asterisk 12, and is now built on the
4259 Stasis message bus. Please see the specification for CEL on the Asterisk
4260 wiki at https://wiki.asterisk.org/wiki/x/4ICLAQ for more detailed
4263 * The 'extra' field of all CEL events that use it now consists of a JSON blob
4264 with key/value pairs which are defined in the Asterisk 12 CEL documentation.
4266 * BLINDTRANSFER events now report the transferee bridge unique
4267 identifier, extension, and context in a JSON blob as the extra string
4268 instead of the transferee channel name as the peer.
4270 * ATTENDEDTRANSFER events now report the peer as NULL and additional
4271 information in the 'extra' string as a JSON blob. For transfers that occur
4272 between two bridged channels, the 'extra' JSON blob contains the primary
4273 bridge unique identifier, the secondary channel name, and the secondary
4274 bridge unique identifier. For transfers that occur between a bridged channel
4275 and a channel running an app, the 'extra' JSON blob contains the primary
4276 bridge unique identifier, the secondary channel name, and the app name.
4278 * LOCAL_OPTIMIZE events have been added to convey local channel
4279 optimizations with the record occurring for the semi-one channel and
4280 the semi-two channel name in the peer field.
4282 * BRIDGE_START, BRIDGE_END, BRIDGE_UPDATE, 3WAY_START, 3WAY_END, CONF_ENTER,
4283 CONF_EXIT, CONF_START, and CONF_END events have all been removed. These
4284 events have been replaced by BRIDGE_ENTER/BRIDGE_EXIT. The BRIDGE_ENTER
4285 and BRIDGE_EXIT events are raised when a channel enters/exits any bridge,
4286 regardless of whether or not that bridge happens to contain multiple
4291 * When compiled with '--enable-dev-mode', the astobj2 library will now add
4292 several CLI commands that allow for inspection of ao2 containers that
4293 register themselves with astobj2. The CLI commands are 'astobj2 container
4294 dump', 'astobj2 container stats', and 'astobj2 container check'.
4296 * Added specific CLI commands for bridge inspection. This includes 'bridge
4297 show all', which lists all bridges in the system, and 'bridge show {id}',
4298 which provides specific information about a bridge.
4300 * Added CLI command 'bridge destroy'. This will destroy the specified bridge,
4301 ejecting the channels currently in the bridge. If the channels cannot
4302 continue in the dialplan or application that put them in the bridge, they
4305 * Added command 'bridge kick'. This will eject a single channel from a bridge.
4307 * Added commands to inspect and manipulate the registered bridge technologies.
4308 This include 'bridge technology show', which lists the registered bridge
4309 technologies, as well as 'bridge technology {suspend|unsuspend} {tech}',
4310 which controls whether or not a registered bridge technology can be used
4311 during smart bridge operations. If a technology is suspended, it will not
4312 be used when a bridge technology is picked for channels; when unsuspended,
4313 it can be used again.
4315 * The command 'config show help {module} {type} {option}' will show
4316 configuration documentation for modules with XML configuration
4317 documentation. When {module}, {type}, and {option} are omitted, a listing
4318 of all modules with registered documentation is displayed. When {module}
4319 is specified, a listing of all configuration types for that module is
4320 displayed, along with their synopsis. When {module} and {type} are
4321 specified, a listing of all configuration options for that type are
4322 displayed along with their synopsis. When {module}, {type}, and {option}
4323 are specified, detailed information for that configuration option is
4326 * Added 'core show sounds' and 'core show sound' CLI commands. These display
4327 a listing of all installed media sounds available on the system and
4328 detailed information about a sound, respectively.
4330 * 'xmldoc dump' has been added. This CLI command will dump the XML
4331 documentation DOM as a string to the specified file. The Asterisk core
4332 will populate certain XML elements pulled from the source files with
4333 additional run-time information; this command lets a user produce the
4334 XML documentation with all information.
4338 * Parking has been pulled from core and placed into a separate module called
4339 res_parking. See Parking changes below for more details. Configuration for
4340 parking should now be performed in res_parking.conf. Configuration for
4341 parking in features.conf is now unsupported.
4343 * Core attended transfers now have several new options. While performing an
4344 attended transfer, the transferer now has the following options:
4345 - *1 - cancel the attended transfer (configurable via atxferabort)
4346 - *2 - complete the attended transfer, dropping out of the call
4347 (configurable via atxfercomplete)
4348 - *3 - complete the attended transfer, but stay in the call. This will turn
4349 the call into a multi-party bridge (configurable via atxferthreeway)
4350 - *4 - swap to the other party. Once an attended transfer has begun, this
4351 options may be used multiple times (configurable via atxferswap)
4353 * For DTMF blind and attended transfers, the channel variable TRANSFER_CONTEXT
4354 must be on the channel initiating the transfer to have any effect.
4356 * The BRIDGE_FEATURES channel variable would previously only set features for
4357 the calling party and would set this feature regardless of whether the
4358 feature was in caps or in lowercase. Use of a caps feature for a letter
4359 will now apply the feature to the calling party while use of a lowercase
4360 letter will apply that feature to the called party.
4362 * Add support for automixmon to the BRIDGE_FEATURES channel variable.
4364 * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is
4365 removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that
4366 activated the dynamic feature.
4368 * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set
4369 only on the channel executing the dynamic feature. Executing a dynamic
4370 feature on the bridge peer in a multi-party bridge will execute it on all
4371 peers of the activating channel.
4373 * You can now have the settings for a channel updated using the FEATURE()
4374 and FEATUREMAP() functions inherited to child channels by setting
4375 FEATURE(inherit)=yes.
4377 * automixmon now supports additional channel variables from automon including:
4378 TOUCH_MIXMONITOR_PREFIX, TOUCH_MIXMONITOR_MESSAGE_START,
4379 and TOUCH_MIXMONITOR_MESSAGE_STOP
4381 * A new general features.conf option 'recordingfailsound' has been added which
4382 allowssetting a failure sound for a user tries to invoke a recording feature
4383 such as automon or automixmon and it fails.
4385 * It is no longer necessary (or possible) to define the ATXFER_NULL_TECH in
4386 features.c for atxferdropcall=no to work properly. This option now just
4391 * Added log rotation strategy 'none'. If set, no log rotation strategy will
4392 be used. Given that this can cause the Asterisk log files to grow quickly,
4393 this option should only be used if an external mechanism for log management
4398 * Dynamic realtime tables for SIP Users can now include a 'path' field. This
4399 will store the path information for that peer when it registers. Realtime
4400 tables can also use the 'supportpath' field to enable Path header support.
4402 * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
4403 objectIdentifier. This maps to the supportpath option in sip.conf.
4407 * Sorcery is a new data abstraction and object persistence API in Asterisk. It
4408 provides modules a useful abstraction on top of the many storage mechanisms
4409 in Asterisk, including the Asterisk Database, static configuration files,
4410 static Realtime, and dynamic Realtime. It also provides a caching service.
4411 Users can configure a hierarchy of data storage layers for specific modules
4414 * All future modules which utilize Sorcery for object persistence must have a
4415 column named "id" within their schema when using the Sorcery realtime module.
4416 This column must be able to contain a string of up to 128 characters in length.
4418 Security Events Framework
4420 * Security Event timestamps now use ISO 8601 formatted date/time instead of
4421 the "seconds-microseconds" format that it was using previously.
4425 * The Stasis message bus is a publish/subscribe message bus internal to
4426 Asterisk. Many services in Asterisk are built on the Stasis message bus,
4427 including AMI, ARI, CDRs, and CEL. Parameters controlling the operation of
4428 Stasis can be configured in stasis.conf. Note that these parameters operate
4429 at a very low level in Asterisk, and generally will not require changes.
4433 * When a channel driver is configured to enable jiterbuffers, they are now
4434 applied unconditionally when a channel joins a bridge. If a jitterbuffer
4435 is already set for that channel when it enters, such as by the JITTERBUFFER
4436 function, then the existing jitterbuffer will be used and the one set by
4437 the channel driver will not be applied.
4441 * chan_agent has been removed and replaced with AgentLogin and AgentRequest
4442 dialplan applications provided by the app_agent_pool module. Agents are
4443 connected with callers using the new AgentRequest dialplan application.
4444 The Agents:<agent-id> device state is available to monitor the status of an
4445 agent. See agents.conf.sample for valid configuration options.
4447 * The updatecdr option has been removed. Altering the names of channels on a
4448 CDR is not supported - the name of the channel is the name of the channel,
4449 and pretending otherwise helps no one. The AGENTUPDATECDR channel variable
4450 has also been removed, for the same reason.
4452 * The endcall and enddtmf configuration options are removed. Use the
4453 dialplan function CHANNEL(dtmf_features) to set DTMF features on the agent
4454 channel before calling AgentLogin.
4458 * chan_bridge has been removed. Its functionality has been incorporated
4459 directly into the ConfBridge application itself.
4463 * Added the CLI command 'pri destroy span'. This will destroy the D-channel
4464 of the specified span and its B-channels. Note that this command should
4465 only be used if you understand the risks it entails.
4467 * The CLI command 'dahdi destroy channel' is now 'dahdi destroy channels'.
4468 A range of channels can be specified to be destroyed. Note that this command
4469 should only be used if you understand the risks it entails.
4471 * Added the CLI command 'dahdi create channels'. A range of channels can be
4472 specified to be created, or the keyword 'new' can be used to add channels
4475 * The script specified by the chan_dahdi.conf mwimonitornotify option now gets
4476 the exact configured mailbox name. For app_voicemail mailboxes this is
4479 * Added mwi_vm_boxes that also must be configured for ISDN MWI to be enabled.
4483 * IPv6 support has been added. We are now able to bind to and
4484 communicate using IPv6 addresses.
4488 * The /b option has been removed.
4490 * chan_local moved into the system core and is no longer a loadable module.
4494 * Added general support for busy detection.
4496 * Added ECAM command support for Sony Ericsson phones.
4500 * A new SIP channel driver for Asterisk, chan_pjsip is built on the PJSIP
4501 SIP stack. A collection of resource modules provides the bulk of the SIP
4502 functionality. For more information on the new SIP channel driver, see
4503 https://wiki.asterisk.org/wiki/x/JYGLAQ
4507 * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
4508 using the 'supportpath' setting, either on a global basis or on a peer basis.
4509 This setting enables Asterisk to route outgoing out-of-dialog requests via a
4510 set of proxies by using a pre-loaded route-set defined by the Path headers in
4511 the REGISTER request. See Realtime updates for more configuration information.
4513 * The SIP_CODEC family of variables may now specify more than one codec. Each
4514 codec must be separated by a comma. The first codec specified is the
4515 preferred codec for the offer. This allows a dialplan writer to specify both
4516 audio and video codecs, e.g., Set(SIP_CODEC=ulaw,h264)
4518 * The 'callevents' parameter has been removed. Hold AMI events are now raised
4519 in the core, and can be filtered out using the 'eventfilter' parameter
4522 * Added 'ignore_requested_pref'. When enabled, this will use the preferred
4523 codecs configured for a peer instead of the requested codec.
4525 * The option "register_retry_403" has been added to chan_sip to work around
4526 servers that are known to erroneously send 403 in response to valid
4527 REGISTER requests and allows Asterisk to continue attepmting to connect.
4531 * Added the 'immeddialkey' parameter. If set, when the user presses the
4532 configured key the already entered number will be immediately dialed. This
4533 is useful when the dialplan allows for variable length pattern matching.
4534 Valid options are '*' and '#'.
4536 * Added the 'callfwdtimeout' parameter. This configures the amount of time (in
4537 milliseconds) before a call forward is considered to not be answered.
4539 * The 'serviceurl' parameter allows Service URLs to be attached to line
4548 * The password option has been disabled, as the AgentLogin application no
4549 longer provides authentication.
4553 * Due to changes in the Asterisk core, this function is no longer needed to
4554 preserve a MixMonitor on a channel during transfer operations and dialplan
4555 execution. It is effectively obsolete.
4559 * The 'amaflags' and 'accountcode' attributes for the CDR function are
4560 deprecated. Use the CHANNEL function instead to access these attributes.
4562 * The 'l' option has been removed. When reading a CDR attribute, the most
4563 recent record is always used. When writing a CDR attribute, all non-finalized
4566 * The 'r' option has been removed, for the same reason as the 'l' option.
4568 * The 's' option has been removed, as LOCKED semantics no longer exist in the
4573 * A new function CDR_PROP has been added. This function lets you set properties
4574 on a channel's active CDRs. This function is write-only. Properties accept
4575 boolean values to set/clear them on the channel's CDRs. Valid properties
4577 - 'party_a' - make this channel the preferred Party A in any CDR between two
4578 channels. If two channels have this property set, the creation time of the
4579 channel is used to determine who is Party A. Note that dialed channels are
4580 never Party A in a CDR.
4581 - 'disable' - disable CDRs on this channel. This is analogous to the NoCDR
4582 application when set to True, and analogous to the 'e' option in ResetCDR
4587 * Added the argument 'dtmf_features'. This sets the DTMF features that will be
4588 enabled on a channel when it enters a bridge. Allowed values are 'T', 'K',
4589 'H', 'W', and 'X', and are analogous to the parameters passed to the Dial
4592 * Added the argument 'after_bridge_goto'. This can be set to a parseable Goto
4593 string, i.e., [[context],extension],priority. If set on a channel, if a
4594 channel leaves a bridge but is not hung up it will resume dialplan execution
4599 * JITTERBUFFER now accepts an argument of 'disabled' which can be used
4600 to remove jitterbuffers previously set on a channel with JITTERBUFFER.
4601 The value of this setting is ignored when disabled is used for the argument.
4605 * A new function provided by chan_pjsip, this function can be used in
4606 conjunction with the Dial application to construct a dial string that will
4607 dial all contacts on an Address of Record associated with a chan_pjsip
4612 * Provided by chan_pjsip, this function sets the codecs to be offered on the
4613 outbound channel prior to dialing.
4617 * Redirecting reasons can now be set to arbitrary strings. This means
4618 that the REDIRECTING dialplan function can be used to set the redirecting
4619 reason to any string. It also allows for custom strings to be read as the
4620 redirecting reason from SIP Diversion headers.
4624 * The SPEECH_ENGINE function now supports read operations. When read from, it
4625 will return the current value of the requested attribute.
4629 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
4630 system as mailbox@context. The rest of the system cannot add @default
4631 to mailbox identifiers for app_voicemail that do not specify a context
4632 any longer. It is a mailbox identifier format that should only be
4633 interpreted by app_voicemail.
4639 res_agi (Asterisk Gateway Interface)
4641 * The manager event AGIExec has been split into AGIExecStart and AGIExecEnd.
4643 * The manager event AsyncAGI has been split into AsyncAGIStart, AsyncAGIExec,
4646 * The CONTROL STREAM FILE command now accepts an offsetms parameter. This
4647 will start the playback of the audio at the position specified. It will
4648 also return the final position of the file in 'endpos'.
4650 * The CONTROL STREAM FILE command will now populate the CPLAYBACKSTATUS
4651 channel variable if the user stopped the file playback or if a remote
4652 entity stopped the playback. If neither stopped the playback, it will
4653 indicate the overall success/failure of the playback. If stopped early,
4654 the final offset of the file will be set in the CPLAYBACKOFFSET channel
4657 * The SAY ALPHA command now accepts an additional parameter to control
4658 whether it specifies the case of uppercase, lowercase, or all letters to
4659 provide functionality similar to SayAlphaCase.
4661 res_ari (Asterisk RESTful Interface) (and others)
4663 * The Asterisk RESTful Interface (ARI) provides a mechanism to expose and
4664 control telephony primitives in Asterisk by remote client. This includes
4665 channels, bridges, endpoints, media, and other fundamental concepts. Users
4666 of ARI can develop their own communications applications, controlling
4667 multiple channels using an HTTP RESTful interface and receiving JSON events
4668 about the objects via a WebSocket connection. ARI can be configured in
4669 Asterisk via ari.conf. For more information on ARI, see
4670 https://wiki.asterisk.org/wiki/x/0YCLAQ
4674 * Parking has been extracted from the Asterisk core as a loadable module,
4675 res_parking. Configuration for parking is now provided by res_parking.conf.
4676 Configuration through features.conf is no longer supported.
4678 * res_parking uses the configuration framework. If an invalid configuration is
4679 supplied, res_parking will fail to load or fail to reload. Previously,
4680 invalid configurations would generally be accepted, with certain errors
4681 resulting in individually disabled parking lots.
4683 * Parked calls are now placed in bridges. While this is largely an
4684 architectural change, it does have implications on how channels in a parking
4685 lot are viewed. For example, commands that display channels in bridges will
4686 now also display the channels in a parking lot.
4688 * The order of arguments for the new parking applications have been modified.
4689 Timeout and return context/exten/priority are now implemented as options,
4690 while the name of the parking lot is now the first parameter. See the
4691 application documentation for Park, ParkedCall, and ParkAndAnnounce for more
4692 in-depth information as well as syntax.
4694 * Extensions are by default no longer automatically created in the dialplan to
4695 park calls or pickup parked calls. Generation of dialplan extensions can be
4696 enabled using the 'parkext' configuration option.
4698 * ADSI functionality for parking is no longer supported. The 'adsipark'
4699 configuration option has been removed as a result.
4701 * The PARKINGSLOT channel variable has been deprecated in favor of
4702 PARKING_SPACE to match the naming scheme of the new system.
4704 * PARKING_SPACE and PARKEDLOT channel variables will now be set for a parked
4705 channel even when the configuration option 'comebactoorigin' is enabled.
4707 * A new CLI command 'parking show' has been added. This allows a user to
4708 inspect the parking lots that are currently in use.
4709 'parking show <parkinglot>' will also show the parked calls in a specific
4712 * The CLI command 'parkedcalls' is now deprecated in favor of
4713 'parking show <parkinglot>'.
4715 * The AMI command 'ParkedCalls' will now accept a 'ParkingLot' argument which
4716 can be used to get a list of parked calls for a specific parking lot.
4718 * The AMI command 'Park' field 'Channel2' has been deprecated and replaced
4719 with 'TimeoutChannel'. If both 'Channel2' and 'TimeoutChannel' are
4720 specified, 'TimeoutChannel' will be used. The field 'TimeoutChannel' is no
4721 longer a required argument.
4723 * The ParkAndAnnounce application is now provided through res_parking instead
4724 of through the separate app_parkandannounce module.
4726 * ParkAndAnnounce will no longer go to the next position in dialplan on timeout
4727 by default. Instead, it will follow the timeout rules of the parking lot. The
4728 old behavior can be reproduced by using the 'c' option.
4730 * Dynamic parking lots will now fail to be created under the following
4732 - if the parking lot specified by PARKINGDYNAMIC does not exist
4733 - if they require exclusive park and parkedcall extensions which overlap
4734 with existing parking lots.
4736 * Dynamic parking lots will be cleared on reload for dynamic parking lots that
4737 currently contain no calls. Dynamic parking lots containing parked calls
4738 will persist through the reloads without alteration.
4740 * If 'parkext_exclusive' is set for a parking lot and that extension is
4741 already in use when that parking lot tries to register it, this is now
4742 considered a parking system configuration error. Configurations which do
4743 this will be rejected.
4745 * Added channel variable PARKER_FLAT. This contains the name of the extension
4746 that would be used if 'comebacktoorigin' is enabled. This can be useful when
4747 comebacktoorigin is disabled, but the dialplan or an external control
4748 mechanism wants to use the extension in the park-dial context that was
4749 generated to re-dial the parker on timeout.
4751 res_pjsip (and many others)
4753 * A large number of resource modules make up the SIP stack based on pjsip.
4754 The chan_pjsip channel driver users these resource modules to provide
4755 various SIP functionality in Asterisk. The majority of configuration for
4756 these modules is performed in pjsip.conf. Other modules may use their
4757 own configuration files.
4759 * Added 'set_var' option for an endpoint. For each variable specified that
4760 variable gets set upon creation of a channel involving the endpoint.
4764 * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
4765 them, an Asterisk-specific version of PJSIP needs to be installed.
4766 Tarballs are available from https://github.com/asterisk/pjproject/tags/.
4768 res_statsd/res_chan_stats
4770 * A new resource module, res_statsd, has been added, which acts as a statsd
4771 client. This module allows Asterisk to publish statistics to a statsd
4772 server. In conjunction with res_chan_stats, it will publish statistics about
4773 channels to the statsd server. It can be configured via res_statsd.conf.
4777 * Device state for XMPP buddies is now available using the following format:
4778 XMPP/<client name>/<buddy address>
4779 If any resource is available the device state is considered to be not in use.
4780 If no resources exist or all are unavailable the device state is considered
4787 Realtime/Database Scripts
4789 * Asterisk previously included example db schemas in the contrib/realtime/
4790 directory of the source tree. This has been replaced by a set of database
4791 migrations using the Alembic framework. This allows you to use alembic to
4792 initialize the database for you. It will also serve as a database migration
4793 tool when upgrading Asterisk in the future.
4795 See contrib/ast-db-manage/README.md for more details.
4799 * A new script has been added in the contrib/scripts/sip_to_res_pjsip folder.
4800 This python script will convert an existing sip.conf file to a
4801 pjsip.conf file, for use with the chan_pjsip channel driver. This script
4802 is meant to be an aid in converting an existing chan_sip configuration to
4803 a chan_pjsip configuration, but it is expected that configuration beyond
4804 what the script provides will be needed.
4806 ------------------------------------------------------------------------------
4807 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
4808 ------------------------------------------------------------------------------
4812 * The Asterisk build system will now build and install a shared library
4813 (libasteriskssl.so) used to wrap various initialization and shutdown functions
4814 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
4815 that Asterisk can ensure that these functions do *not* get called by any
4816 modules that are loaded into Asterisk, since they should only be called once
4817 in any single process. If desired, this feature can be disabled by supplying
4818 the "--disable-asteriskssl" option to the configure script.
4820 * A new make target, 'full', has been added to the Makefile. This performs
4821 the same compilation actions as make all, but will also scan the entirety of
4822 each source file for documentation. This option is needed to generate AMI
4823 event documentation. Note that your system must have Python in order for
4824 this make target to succeed.
4826 * The optimization portion of the build system has been reworked to avoid
4827 broken builds on certain architectures. All architecture-specific
4828 optimization has been removed in favor of using -march=native to allow gcc
4829 to detect the environment in which it is running when possible. This can
4830 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
4832 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
4833 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
4835 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
4836 previously parsed the header file to obtain the version of Asterisk, you
4837 will now have to go through Asterisk to get the version information.
4845 * Added 'F()' option. Similar to the dial option, this can be supplied with
4846 arguments indicating where the callee should go after the caller is hung up,
4847 or without options specified, the priority after the Queue will be used.
4852 * Added menu action admin_toggle_mute_participants. This will mute / unmute
4853 all non-admin participants on a conference. The confbridge configuration
4854 file also allows for the default sounds played to all conference users when
4855 this occurs to be overriden using sound_participants_unmuted and
4856 sound_participants_muted.
4858 * Added menu action participant_count. This will playback the number of
4859 current participants in a conference.
4861 * Added announcement configuration option to user profile. If set the sound
4862 file will be played to the user, and only the user, upon joining the
4865 * Added record_file_append option that defaults to "yes", but if set to no
4866 will create a new file between each start/stop recording.
4871 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
4872 channels respectively before the callee channels are called.
4877 * Added support for IPv6.
4879 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
4880 external process will cause the current playlist to be cleared, including
4881 stopping any audio file that is currently playing. This is useful when you
4882 want to interrupt audio playback only when specific DTMF is entered by the
4888 * A new option, 'I' has been added to app_followme. By setting this option,
4889 Asterisk will not update the caller with connected line changes when they
4890 occur. This is similar to app_dial and app_queue.
4892 * The 'N' option is now ignored if the call is already answered.
4894 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
4895 and caller channels respectively before the callee channels are called.
4897 * The winning FollowMe outgoing call is now put on hold if the caller put it on
4903 * MixMonitor hooks now have IDs associated with them which can be used to
4904 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
4905 will allow storage of the MixMonitor ID in a channel variable. StopMixmonitor
4906 now accepts that ID as an argument.
4908 * Added 'm' option, which stores a copy of the recording as a voicemail in the
4909 indicated mailboxes.
4914 * The connect action in app_mysql now allows you to specify a port number to
4915 connect to. This is useful if you run a MySQL server on a non-standard
4921 * Increased the default number of allowed destinations from 5 to 12.
4926 * The app_page application now no longer depends on DAHDI or app_meetme. It
4927 has been re-architected to use app_confbridge internally.
4932 * Added queue options autopausebusy and autopauseunavail for automatically
4933 pausing a queue member when their device reports busy or congestion.
4935 * The 'ignorebusy' option for queue members has been deprecated in favor of
4936 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
4937 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
4938 per interface basis. Individual ringinuse values can now be set in
4939 queues.conf via an argument to member definitions. Lastly, the queue
4940 'ringinuse' setting now only determines defaults for the per member
4941 'ringinuse' setting and does not override per member settings like it does
4942 in earlier versions.
4944 * Added 'F()' option. Similar to the dial option, this can be supplied with
4945 arguments indicating where the callee should go after the caller is hung up,
4946 or without options specified, the priority after the Queue will be used.
4948 * Added new option log_member_name_as_agent, which will cause the membername to
4949 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
4950 state_interface has been set.
4952 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
4954 * App_queue will now play periodic announcements for the caller that
4955 holds the first position in the queue&nb
projects / asterisk/asterisk.git / blob