1 ==============================================================================
3 === This file documents the new and/or enhanced functionality added in
4 === the Asterisk versions listed below. This file does NOT include
5 === changes in behavior that would not be backwards compatible with
6 === previous versions; for that information see the UPGRADE.txt file
7 === and the other UPGRADE files for older releases.
9 ==============================================================================
10 ------------------------------------------------------------------------------
11 --- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
12 ------------------------------------------------------------------------------
15 AMI (Asterisk Manager Interface)
17 * The SIPshowpeer action will now include a 'SubscribeContext' field for a peer
18 in its response if the peer has a subscribe context set.
20 * The SIPqualifypeer action now acknowledges the request once it has established
21 that the request is against a known peer. It also issues a new event,
22 'SIPqualifypeerdone', once the qualify action has been completed.
24 * The PlayDTMF action now supports an optional 'Duration' parameter. This
25 specifies the duration of the digit to be played, in milliseconds.
27 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
28 updates when changes occur instead of requiring the use of pollmailboxes.
30 * CLI Command 'Manager Show Commands' no longer truncates command names longer
31 than 15 characters and no longer shows authorization requirement for commands.
32 'Manager Show Command' now displays the privileges needed for using a given
33 manager command instead.
35 * Added new action "ControlPlayback". The ControlPlayback action allows an AMI
36 client to manipulate audio currently being played back on a channel. The
37 supported operations depend on the application being used to send audio to
38 the channel. When the audio playback was initiated using the ControlPlayback
39 application or CONTROL STREAM FILE AGI command, the audio can be paused,
40 stopped, restarted, reversed, or skipped forward. When initiated by other
41 mechanisms (such as the Playback application), the audio can be stopped,
42 reversed, or skipped forward.
44 * Channel related events now contain a snapshot of channel state, adding new
45 fields to many of these events.
47 * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
48 in a future release. Please use the common 'Exten' field instead.
50 * The AMI event 'UserEvent' from app_userevent now contains the channel state
51 fields. The channel state fields will come before the body fields.
53 * The deprecated use of | (pipe) as a separator in the channelvars setting in
54 manager.conf has been removed.
56 * Channel Variables conveyed with a channel no longer contain the name of the
57 channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now
58 ChanVariable: bar=baz. When multiple channels are present in a single AMI
59 event, the various ChanVariable fields will contain a suffix that specifies
60 which channel they correspond to.
67 * Added general support for busy detection.
69 * Added ECAM command support for Sony Ericsson phones.
73 * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
74 using the 'supportpath' setting, either on a global basis or on a peer basis.
75 This setting enables Asterisk to route outgoing out-of-dialog requests via a
76 set of proxies by using a pre-loaded route-set defined by the Path headers in
77 the REGISTER request. See Realtime updates for more configuration information.
81 * The BRIDGE_FEATURES channel variable would previously only set features for
82 the calling party and would set this feature regardless of whether the
83 feature was in caps or in lowercase. Use of a caps feature for a letter
84 will now apply the feature to the calling party while use of a lowercase
85 letter will apply that feature to the called party.
87 * Add support for automixmonitor to the BRIDGE_FEATURES channel variable.
89 * PARKINGSLOT and PARKEDLOT channel variables will now be set for a parked
90 channel even when comebactoorigin=yes
94 * When performing queue pause/unpause on an interface without specifying an
95 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
96 least one member of any queue exists for that interface.
98 * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
99 for realtime queue log entries.
103 * Added the 'n' option to MeetMe to prevent application of the DENOISE function
104 to a channel joining a conference. Some channel drivers that vary the number
105 of audio samples in a voice frame will experience significant quality problems
106 if a denoiser is attached to the channel; this option gives them the ability
107 to remove the denoiser without having to unload func_speex.
111 * Add queue available hint. exten => 8501,hint,Queue:markq_avail
112 Note: the suffix '_avail' after the queuename.
113 Reports 'InUse' for no logged in agents or no free agents.
114 Reports 'Idle' when an agent is free.
118 * Redirecting reasons can now be set to arbitrary strings. This means
119 that the REDIRECTING dialplan function can be used to set the redirecting
120 reason to any string. It also allows for custom strings to be read as the
121 redirecting reason from SIP Diversion headers.
125 * Dynamic realtime tables for SIP Users can now include a 'path' field. This
126 will store the path information for that peer when it registers. Realtime
127 tables can also use the 'supportpath' field to enable Path header support.
129 * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
130 objectIdentifier. This maps to the supportpath option in sip.conf.
134 * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
135 them, an Asterisk-specific version of pjproject needs to be installed.
136 Tarballs are available from https://github.com/asterisk/pjproject/tags/.
140 * Device state for XMPP buddies is now available using the following format:
141 XMPP/<client name>/<buddy address>
142 If any resource is available the device state is considered to be not in use.
143 If no resources exist or all are unavailable the device state is considered
146 ------------------------------------------------------------------------------
147 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
148 ------------------------------------------------------------------------------
152 * The Asterisk build system will now build and install a shared library
153 (libasteriskssl.so) used to wrap various initialization and shutdown functions
154 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
155 that Asterisk can ensure that these functions do *not* get called by any
156 modules that are loaded into Asterisk, since they should only be called once
157 in any single process. If desired, this feature can be disabled by supplying
158 the "--disable-asteriskssl" option to the configure script.
160 * A new make target, 'full', has been added to the Makefile. This performs
161 the same compilation actions as make all, but will also scan the entirety of
162 each source file for documentation. This option is needed to generate AMI
163 event documentation. Note that your system must have Python in order for
164 this make target to succeed.
166 * The optimization portion of the build system has been reworked to avoid
167 broken builds on certain architectures. All architecture-specific
168 optimization has been removed in favor of using -march=native to allow gcc
169 to detect the environment in which it is running when possible. This can
170 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
172 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
173 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
175 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
176 previously parsed the header file to obtain the version of Asterisk, you
177 will now have to go through Asterisk to get the version information.
185 * Added 'F()' option. Similar to the dial option, this can be supplied with
186 arguments indicating where the callee should go after the caller is hung up,
187 or without options specified, the priority after the Queue will be used.
192 * Added menu action admin_toggle_mute_participants. This will mute / unmute
193 all non-admin participants on a conference. The confbridge configuration
194 file also allows for the default sounds played to all conference users when
195 this occurs to be overriden using sound_participants_unmuted and
196 sound_participants_muted.
198 * Added menu action participant_count. This will playback the number of
199 current participants in a conference.
201 * Added announcement configuration option to user profile. If set the sound
202 file will be played to the user, and only the user, upon joining the
205 * Added record_file_append option that defaults to "yes", but if set to no
206 will create a new file between each start/stop recording.
211 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
212 channels respectively before the callee channels are called.
217 * Added support for IPv6.
219 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
220 external process will cause the current playlist to be cleared, including
221 stopping any audio file that is currently playing. This is useful when you
222 want to interrupt audio playback only when specific DTMF is entered by the
228 * A new option, 'I' has been added to app_followme. By setting this option,
229 Asterisk will not update the caller with connected line changes when they
230 occur. This is similar to app_dial and app_queue.
232 * The 'N' option is now ignored if the call is already answered.
234 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
235 and caller channels respectively before the callee channels are called.
237 * The winning FollowMe outgoing call is now put on hold if the caller put it on
243 * MixMonitor hooks now have IDs associated with them which can be used to
244 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
245 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
246 now accepts that ID as an argument.
248 * Added 'm' option, which stores a copy of the recording as a voicemail in the
254 * The connect action in app_mysql now allows you to specify a port number to
255 connect to. This is useful if you run a MySQL server on a non-standard
261 * Increased the default number of allowed destinations from 5 to 12.
266 * The app_page application now no longer depends on DAHDI or app_meetme. It
267 has been re-architected to use app_confbridge internally.
272 * Added queue options autopausebusy and autopauseunavail for automatically
273 pausing a queue member when their device reports busy or congestion.
275 * The 'ignorebusy' option for queue members has been deprecated in favor of
276 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
277 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
278 per interface basis. Individual ringinuse values can now be set in
279 queues.conf via an argument to member definitions. Lastly, the queue
280 'ringinuse' setting now only determines defaults for the per member
281 'ringinuse' setting and does not override per member settings like it does
284 * Added 'F()' option. Similar to the dial option, this can be supplied with
285 arguments indicating where the callee should go after the caller is hung up,
286 or without options specified, the priority after the Queue will be used.
288 * Added new option log_member_name_as_agent, which will cause the membername to
289 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
290 state_interface has been set.
292 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
296 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
297 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
298 changed arguments to SayUnixTime so that every option is truly optional even
299 when using multiple options (so that j option could be used without having to
300 manually specify timezone and format) There are other benefits, e.g., format
301 can now be used without specifying time zone as well.
306 * Addition of the VM_INFO function - see Function changes.
308 * The imapserver, imapport, and imapflags configuration options can now be
309 overriden on a user by user basis.
311 * When voicemail plays a message's envelope with saycid set to yes, when
312 reaching the caller id field it will play a recording of a file with the same
313 base name as the sender's callerid if there is a similarly named file in
314 <astspooldir>/recordings/callerids/
316 * Voicemails now contains a unique message identifier "msg_id", which is stored
317 in the message envelope with the sound files. IMAP backends will now store
318 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
319 backends will store the message identifier in a "msg_id" column. See
320 UPGRADE.txt for more information.
322 * Added VoiceMailPlayMsg application. This application will play a single
323 voicemail message from a mailbox. The result of the application, SUCCESS or
324 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
329 * Hangup handlers can be attached to channels using the CHANNEL() function.
330 Hangup handlers will run when the channel is hung up similar to the h
331 extension. The hangup_handler_push option will push a GoSub compatible
332 location in the dialplan onto the channel's hangup handler stack. The
333 hangup_handler_pop option will remove the last added location, and optionally
334 replace it with a new GoSub compatible location. The hangup_handler_wipe
335 option will remove all locations on the stack, and optionally add a new
338 * The expression parser now recognizes the ABS() absolute value function,
339 which will convert negative floating point values to positive values.
341 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
342 control of faxdetect.
344 * Addition of the VM_INFO function that can be used to retrieve voicemail
345 user information, such as the email address and full name.
346 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
349 * The REDIRECTING function now supports the redirecting original party id
352 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
353 lets you set some of the configuration options from the [general] section
354 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
355 the key sequence used to activate built-in features, such as blindxfer,
356 and automon. See the built-in documentation for details.
358 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
359 instead of simply the uri. This is the format that MessageSend() can use
360 in the from parameter for outgoing SIP messages.
362 * Added the PRESENCE_STATE function. This allows retrieving presence state
363 information from any presence state provider. It also allows setting
364 presence state information from a CustomPresence presence state provider.
365 See AMI/CLI changes for related commands.
367 * Added the AMI_CLIENT function to make manager account attributes available
368 to the dialplan. It currently supports returning the current number of
369 active sessions for a given account.
371 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
372 and the REDIRECTING functions.
380 * Added a manager event "LocalBridge" for local channel call bridges between
381 the two pseudo-channels created.
386 * Added dialtone_detect option for analog ports to disconnect incoming
387 calls when dialtone is detected.
389 * Added option colp_send to send ISDN connected line information. Allowed
390 settings are block, to not send any connected line information; connect, to
391 send connected line information on initial connect; and update, to send
392 information on any update during a call. Default is update.
394 * Add options namedcallgroup and namedpickupgroup to support installations
395 where a higher number of groups (>64) is required.
397 * Added support to use private party ID information with PRI calls.
402 * A new channel driver named chan_motif has been added which provides support for
403 Google Talk and Jingle in a single channel driver. This new channel driver includes
404 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
405 hold, unhold, and ringing notification. It is also compliant with the current Jingle
406 specification, current Google Jingle specification, and the original Google Talk
412 * Added NAT support for RTP. Setting in config is 'nat', which can be set
413 globally and overriden on a peer by peer basis.
415 * Direct media functionality has been added. Options in config are:
416 directmedia (directrtp) and directrtpsetup (earlydirect)
418 * ChannelUpdate events now contain a CallRef header.
423 * Asterisk will no longer substitute CID number for CID name in the display
424 name field if CID number exists without a CID name. This change improves
425 compatibility with certain device features such as Avaya IP500's directory
428 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
429 created using that setting to not be removed during SIP reload.
431 * Added settings recordonfeature and recordofffeature. When receiving an INFO
432 request with a "Record:" header, this will turn the requested feature on/off.
433 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
434 dynamic features must be enabled and configured properly on the requesting
435 channel for this to function properly.
437 * Add support to realtime for the 'callbackextension' option.
439 * When multiple peers exist with the same address, but differing
440 callbackextension options, incoming requests that are matched by address
441 will be matched to the peer with the matching callbackextension if it is
444 * Two new NAT options, auto_force_rport and auto_comedia, have been added
445 which set the force_rport and comedia options automatically if Asterisk
446 detects that an incoming SIP request crossed a NAT after being sent by
449 * The default global nat setting in sip.conf has been changed from force_rport
452 * NAT settings are now a combinable list of options. The equivalent of the
453 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
455 * Adds an option send_diversion which can be disabled to prevent
456 diversion headers from automatically being added to INVITE requests.
458 * Add support for lightweight NAT keepalive. If enabled a blank packet will
459 be sent to the remote host at a given interval to keep the NAT mapping open.
460 This can be enabled using the keepalive configuration option.
462 * Add option 'tonezone' to specify country code for indications. This option
463 can be set both globally and overridden for specific peers.
465 * The SIP Security Events Framework now supports IPv6.
467 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
468 between multiple user agents. When set, for directmedia reinvites,
469 Asterisk will not send an immediate reinvite on an incoming call leg. This
470 option is useful when peered with another SIP user agent that is known to
471 send immediate direct media reinvites upon call establishment.
473 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
476 * Add options subminexpiry and submaxexpiry to set limits of subscription
477 timer independently from registration timer settings. The setting of the
478 registration timer limits still is done by options minexpiry, maxexpiry
479 and defaultexpiry. For backwards compatibility the setting of minexpiry
480 and maxexpiry also is used to configure the subscription timer limits if
481 subminexpiry and submaxexpiry are not set in sip.conf.
483 * Set registration timer limits to default values when reloading sip
484 configuration and values are not set by configuration.
486 * Add options namedcallgroup and namedpickupgroup to support installations
487 where a higher number of groups (>64) is required.
489 * When a MESSAGE request is received, the address the request was received from
490 is now saved in the SIP_RECVADDR variable.
492 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
493 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
494 the ANI2/OLI information is set on the channel, which can be retrieved using
495 the CALLERID function.
497 * Peers can now be configured to support negotiation of ICE candidates using
498 the setting icesupport. See res_rtp_asterisk changes for more information.
500 * Added support for format attribute negotiation. See the Codecs changes for
503 * Extra headers specified with SIPAddHeader are sent with the REFER message
504 when using Transfer application. See refer_addheaders in sip.conf.sample.
506 * Added support to use private party ID information with calls.
508 * Adds an option discard_remote_hold_retrieval that when set stops telling
509 the peer to start music on hold.
514 * Added skinny version 17 protocol support.
519 * Added ability to use multiple lines for a single phone. This allows multiple
520 calls to occur on a single phone, using callwaiting and switching between calls.
522 * Added option 'sharpdial' allowing end dialing by pressing # key
524 * Added option 'interdigit_timer' to control phone dial timeout
526 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
528 * Added global 'debug' option, that enables debug in channel driver
530 * Added ability to translate on-screen menu in multiple languages. Tested on
531 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
532 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
535 * In addition to English added French and Russian languages for on-screen menus
537 * Reworked dialing number input: added dialing by timeout, immediate dial on
538 on dialplan compare, phone number length now not limited by screen size
540 * Added ability to pickup a call using features.conf defined value and
546 * Add options namedcallgroup and namedpickupgroup to support installations
547 where a higher number of groups (>64) is required.
549 * Added support to use private party ID information with calls.
554 * The minimum DTMF duration can now be configured in asterisk.conf
555 as "mindtmfduration". The default value is (as before) set to 80 ms.
556 (previously it was only available in source code)
558 * Named ACLs can now be specified in acl.conf and used in configurations that
559 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
560 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
561 working ACL. In addition, some CLI commands have been added to provide
562 show information and allow for module reloading - see CLI Changes.
564 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
565 items (separated by commas), and items in the rule can be negated by prefixing
566 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
567 longer necessray to control the order that the 'permit' and 'deny' columns are
568 returned from queries.
570 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
571 be used within the dynamic weight attribute when specifying a mapping.
573 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
574 header, instead of putting the user defined event name there. When enabled
575 the UserDefType header is added for user defined events. This feature is
576 enabled with the setting show_user_defined.
578 * Macro has been deprecated in favor of GoSub. For redirecting and connected
579 line purposes use the following variables instead of their macro equivalents:
580 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
581 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
582 cc_callback_macro in channel configurations.
584 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
587 * Call files now support the "early_media" option to connect with an outgoing
588 extension when early media is received.
590 * Added support to use private party ID information with calls.
595 * A new channel variable, AGIEXITONHANGUP, has been added which allows
596 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
597 AGI application would exit immediately after a channel hangup is detected.
599 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
600 are resolved and each address is attempted in turn until one succeeds or
604 AMI (Asterisk Manager Interface)
606 * The originate action now has an option "EarlyMedia" that enables the
607 call to bridge when we get early media in the call. Previously,
608 early media was disregarded always when originating calls using AMI.
610 * Added setvar= option to manager accounts (much like sip.conf)
612 * Originate now generates an error response if the extension given is not found
615 * MixMonitor will now show IDs associated with the mixmonitor upon creating
616 them if the i(variable) option is used. StopMixMonitor will accept
617 MixMonitorID as an option to close specific MixMonitors.
619 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
620 updated to include information about peers configured with
621 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
622 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
623 returned if auto_force_rport is not enabled.
625 * Added SIPpeerstatus manager command which will generate PeerStatus events
626 similar to the existing PeerStatus events found in chan_sip on demand.
628 * Hangup now can take a regular expression as the Channel option. If you want
629 to hangup multiple channels, use /regex/ as the Channel option. Existing
630 behavior to hanging up a single channel is unchanged, but if you pass a regex,
631 the manager will send you a list of channels back that were hung up.
633 * Support for IPv6 addresses has been added.
635 * AMI Events can now be documented in the Asterisk source. Note that AMI event
636 documentation is only generated when Asterisk is compiled using 'make full'.
637 See the CLI section for commands to display AMI event information.
639 * The AMI Hangup event now includes the AccountCode header so you can easily
640 correlate with AMI Newchannel events.
642 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
643 the StateInterface of the queue member.
645 * Added AMI event SessionTimeout in the Call category that is issued when a
646 call is terminated due to either RTP stream inactivity or SIP session timer
649 * CEL events can now contain a user defined header UserDefType. See core
650 changes for more information.
652 * OOH323 ChannelUpdate events now contain a CallRef header.
654 * Added PresenceState command. This command will report the presence state for
655 the given presence provider.
657 * Added Parkinglots command. This will list all parking lots as a series of
658 AMI Parkinglot events.
660 * Added MessageSend command. This behaves in the same manner as the
661 MessageSend application, and is a technolgoy agnostic mechanism to send out
662 of call text messages.
664 * Added "message" class authorization. This grants an account permission to
665 send out of call messages. Write-only.
670 * The "dialplan add include" command has been modified to create context a context
671 if one does not already exist. For instance, "dialplan add include foo into bar"
672 will create context "bar" if it does not already exist.
674 * A "dialplan remove context" command has been added to remove a context from
677 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
678 filenames of all running mixmonitors on a channel.
680 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
681 numeric instead of 0, 1, or 2.
683 * "stun show status" will show a table describing how the STUN client is
686 * "acl show [named acl]" will show information regarding a Named ACL. The
687 acl module can be reloaded with "reload acl".
689 * Added CLI command to display AMI event information - "manager show events",
690 which shows a list of all known and documented AMI events, and "manager show
691 event [event name]", which shows detail information about a specific AMI
694 * The result of the CLI command "queue show" now includes the state interface
695 information of the queue member.
697 * The command "core set verbose" will now set a separate level of logging for
698 each remote console without affecting any other console.
700 * Added command "cdr show pgsql status" to check connection status
702 * "sip show channel" will now display the complete route set.
704 * Added "presencestate list" command. This command will list all custom
705 presence states that have been set by using the PRESENCE_STATE dialplan
708 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
709 command. This changes a custom presence to a new state.
714 * Codec lists may now be modified by the '!' character, to allow succinct
715 specification of a list of codecs allowed and disallowed, without the
716 requirement to use two different keywords. For example, to specify all
717 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
719 * Add support for parsing SDP attributes, generating SDP attributes, and
720 passing it through. This support includes codecs such as H.263, H.264, SILK,
721 and CELT. You are able to set up a call and have attribute information pass.
722 This should help considerably with video calls.
724 * The iLBC codec can now use a system-provided iLBC library if one is installed,
725 just like the GSM codec.
729 * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
730 'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
734 * Asterisk version and build information is now logged at the beginning of a
737 * Threads belonging to a particular call are now linked with callids which get
738 added to any log messages produced by those threads. Log messages can now be
739 easily identified as involved with a certain call by looking at their call id.
740 Call ids may also be attached to log messages for just about any case where
741 it can be determined to be related to a particular call.
743 * Each logging destination and console now have an independent notion of the
744 current verbosity level. Logger.conf now allows an optional argument to
745 the 'verbose' specifier, indicating the level of verbosity sent to that
746 particular logging destination. Additionally, remote consoles now each
747 have their own verbosity level. The command 'core set verbose' will now set
748 a separate level for each remote console without affecting any other
754 * Added 'announcement' option which will play at the start of MOH and between
755 songs in modes of MOH that can detect transitions between songs (eg.
761 * New per parking lot options: comebackcontext and comebackdialtime. See
762 configs/features.conf.sample for more details.
764 * Channel variable PARKER is now set when comebacktoorigin is disabled in
767 * Channel variable PARKEDCALL is now set with the name of the parking lot
768 when a timeout occurs.
774 CDR Postgresql Driver
776 * Added command "cdr show pgsql status" to check connection status
779 CDR Adaptive ODBC Driver
781 * Added schema option for databases that support specifying a schema.
789 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
790 CALENDAR_WRITE has completed successfully.
795 * A new option, 'probation' has been added to rtp.conf
796 RTP in strictrtp mode can now require more than 1 packet to exit learning
797 mode with a new source (and by default requires 4). The probation option
798 allows the user to change the required number of packets in sequence to any
799 desired value. Use a value of 1 to essentially restore the old behavior.
800 Also, with strictrtp on, Asterisk will now drop all packets until learning
801 mode has successfully exited. These changes are based on how pjmedia handles
802 media sources and source changes.
804 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
805 enabled or disabled using the icesupport setting. A variety of other
806 settings have been introduced to configure STUN/TURN connections.
811 * A new module, res_corosync, has been introduced. This module uses the
812 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
813 of Asterisk servers to both Message Waiting Indication (MWI) and/or
814 Device State (presence) information. This module is very similar to, and
815 is a replacement for the res_ais module that was in previous releases of
821 * This module adds a cleaned up, drop-in replacement for res_jabber called
822 res_xmpp. This provides the same externally facing functionality but is
823 implemented differently internally. res_jabber has been deprecated in favor
824 of res_xmpp; please see the UPGRADE.txt file for more information.
829 * The safe_asterisk script has been updated to allow several of its parameters
830 to be set from environment variables. This also enables a custom run
831 directory of Asterisk to be specified, instead of defaulting to /tmp.
833 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
834 its value to determine the directory to assume is the top-level directory of
835 the source tree. If the variable is not set, it defaults to the current
836 behavior and uses the current working directory.
838 ------------------------------------------------------------------------------
839 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
840 ------------------------------------------------------------------------------
844 * Asterisk now has protocol independent support for processing text messages
845 outside of a call. Messages are routed through the Asterisk dialplan.
846 SIP MESSAGE and XMPP are currently supported. There are options in
847 jabber.conf and sip.conf to allow enabling these features.
848 -> jabber.conf: see the "sendtodialplan" and "context" options.
849 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
850 and "outofcall_message_context" options.
851 The MESSAGE() dialplan function and MessageSend() application have been
852 added to go along with this functionality. More detailed usage information
853 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
854 * If real-time text support (T.140) is negotiated, it will be preferred for
855 sending text via the SendText application. For example, via SIP, messages
856 that were once sent via the SIP MESSAGE request would be sent via RTP if
857 T.140 text is negotiated for a call.
861 * parkedmusicclass can now be set for non-default parking lots.
863 Asterisk Manager Interface
864 --------------------------
865 * PeerStatus now includes Address and Port.
866 * Added Hold events for when the remote party puts the call on and off hold
867 for chan_dahdi ISDN channels.
868 * Added new action MeetmeListRooms to list active conferences (shows same
869 data as "meetme list" at the CLI).
870 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
871 Description field that is set by 'description' in the channel configuration
873 * Added Uniqueid header to UserEvent.
874 * Added new action FilterAdd to control event filters for the current session.
875 This requires the system permission and uses the same filter syntax as
876 filters that can be defined in manager.conf
877 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
878 versions had some instances of the event converted, but others were left
879 as-is. All Unlink events should now be converted to Bridge events. The AMI
880 protocol version number was incremented to 1.2 as a result of this change.
883 --------------------------
884 * The HTTP Server can bind to IPv6 addresses.
887 --------------------------
888 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
889 with busydetect. usage example: busypattern=200,200,200,600
892 --------------------------
893 * New 'gtalk show settings' command showing the current settings loaded from
895 * The 'logger reload' command now supports an optional argument, specifying an
896 alternate configuration file to use.
897 * 'dialplan add extension' command will now automatically create a context if
898 the specified context does not exist with a message indicated it did so.
899 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
900 Description field which can be populated with 'description' in the channel
901 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
904 --------------------------
905 * The filter option in cdr_adaptive_odbc now supports negating the argument,
906 thus allowing records which do NOT match the specified filter.
907 * Added ability to log CONGESTION calls to CDR
910 --------------------------
911 * Ability to define custom SILK formats in codecs.conf.
912 * Addition of speex32 audio format with translation.
913 * CELT codec pass-through support and ability to define
914 custom CELT formats in codecs.conf.
915 * Ability to read raw signed linear files with sample rates
916 ranging from 8khz - 192khz. The new file extensions introduced
917 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
918 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
919 Skinny, H.323, etc) can still only support the following codecs:
920 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
921 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
922 Video: h261, h263, h263p, h264, mpeg4
927 --------------------------
928 * New highly optimized and customizable ConfBridge application capable of
929 mixing audio at sample rates ranging from 8khz-96khz.
930 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
931 and bridge profiles on a channel.
932 * CONFBRIDGE_INFO dialplan function capable of retrieving information
933 about a conference such as locked status and number of parties, admins,
935 * Addition of video_mode option in confbridge.conf for adding video support
936 into a bridge profile.
937 * Addition of the follow_talker video_mode in confbridge.conf. This video
938 mode dynamically switches the video feed to always display the loudest talker
939 supplying video in the conference.
943 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
944 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
945 variables from asterisk.conf.
949 * Addition of the JITTERBUFFER dialplan function. This function allows
950 for jitterbuffering to occur on the read side of a channel. By using
951 this function conference applications such as ConfBridge and MeetMe can
952 have the rx streams jitterbuffered before conference mixing occurs.
953 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
955 * Added STRREPLACE function. This function let's the user search a variable
956 for a given string to replace with another string as many times as the
957 user specifies or just throughout the whole string.
958 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
959 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
960 * Added extensions to chan_ooh323 in function CHANNEL()
962 libpri channel driver (chan_dahdi) DAHDI changes
963 --------------------------
964 * Added moh_signaling option to specify what to do when the channel's bridged
965 peer puts the ISDN channel on hold.
966 * Added display_send and display_receive options to control how the display ie
967 is handled. To send display text from the dialplan use the SendText()
968 application when the option is enabled.
969 * Added mcid_send option to allow sending a MCID request on a span.
972 --------------------------
973 * Added setvar option to calendar.conf to allow setting channel variables on
974 notification channels.
975 * Added "calendar show types" CLI command to list registered calendar
979 --------------------------
980 * Added two new options, r and t with file name arguments to record
981 single direction (unmixed) audio recording separate from the bidirectional
982 (mixed) recording. The mixed file name argument is optional now as long
983 as at least one recording option is used.
986 --------------------------
987 * Added a new option, l, which will disable local call optimization for
988 channels involved with the FollowMe thread. Use this option to improve
989 compatability for a FollowMe call with certain dialplan apps, options, and
993 --------------------------
994 * Added option "k" that will automatically close the conference when there's
995 only one person left when a user exits the conference.
998 --------------------------
999 * cel_pgsql now supports the 'extra' column for data added using the
1000 CELGenUserEvent() application.
1003 --------------------------
1004 * Support for defining hints has been added to pbx_lua. See the 'hints' table
1005 in the sample extensions.lua file for syntax details.
1006 * Applications that perform jumps in the dialplan such as Goto will now
1007 execute properly. When pbx_lua detects that the context, extension, or
1008 priority we are executing on has changed it will immediately return control
1009 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
1010 the priority after the currently executing priority.
1011 * An autoservice is now started by default for pbx_lua channels. It can be
1012 stopped and restarted using the autoservice_stop() and autoservice_start()
1016 --------------------------
1017 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
1018 into a FAXStatus event with an 'Operation' header that will be either
1019 'send', 'receive', and 'gateway'.
1020 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
1021 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
1022 feature will handle converting a fax call between an audio T.30 fax terminal
1023 and an IFP T.38 fax terminal.
1027 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
1028 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
1029 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
1033 * Added general option negative_penalty_invalid default off. when set
1034 members are seen as invalid/logged out when there penalty is negative.
1035 for realtime members when set remove from queue will set penalty to -1.
1036 * Added queue option autopausedelay when autopause is enabled it will be
1037 delayed for this number of seconds since last successful call if there
1038 was no prior call the agent will be autopaused immediately.
1039 * Added member option ignorebusy this when set and ringinuse is not
1040 will allow per member control of multiple calls as ringinuse does for
1045 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
1047 * Added 'k' option to MeetMe to automatically kill the conference when there's only
1048 one participant left (much like a normal call bridge)
1049 * Added extra argument to Originate to set timeout.
1053 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
1054 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
1055 utility in the UTILS section of menuselect. If an existing astdb is found and no
1056 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
1057 convert an existing astdb to the SQLite3 version automatically at runtime.
1061 * Modules marked as deprecated are no longer marked as building by default. Enabling
1062 these modules is still available via menuselect.
1066 * authdebug is now disabled by default. To enable this functionaility again
1067 set authdebug = yes in iax.conf.
1071 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
1072 releases it was disabled.
1076 * The PBX core previously made a call with a non-existing extension test for
1077 extension s@default and jump there if the extension existed.
1078 This was a bad default behaviour and violated the principle of least surprise.
1079 It has therefore been changed in this release. It may affect some
1080 applications and configurations that rely on this behaviour. Most channel
1081 drivers have avoided this for many releases by testing whether the extension
1082 called exists before starting the PBX and generating a local error.
1083 This behaviour still exists and works as before.
1085 Extension "s" is used when no extension is given in a channel driver,
1086 like immediate answer in DAHDI or calling to a domain with no user part
1089 ------------------------------------------------------------------------------
1090 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
1091 ------------------------------------------------------------------------------
1095 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
1096 now defaults to force_rport. It is very important that phones requiring nat=no be
1097 specifically set as such instead of relying on the default setting. If at all
1098 possible, all devices should have nat settings configured in the general section as
1099 opposed to configuring nat per-device.
1100 * Added preferred_codec_only option in sip.conf. This feature limits the joint
1101 codecs sent in response to an INVITE to the single most preferred codec.
1102 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
1103 to be used for the outgoing call. It must be one of the codecs configured
1105 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
1106 to be used for holding a private key. If tlsprivatekey is not specified,
1107 tlscertfile is searched for both public and private key.
1108 * Added tlsclientmethod option to sip.conf. This allows the protocol for
1109 outbound client connections to be specified.
1110 * The sendrpid parameter has been expanded to include the options
1111 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
1112 header to be sent (equivalent to setting sendrpid=yes) and setting
1113 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
1114 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
1115 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
1116 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
1117 will accept the SDP even if the SDP version number is not properly incremented,
1118 but will generate a warning in the log indicating that the SIP peer that sent
1119 the SDP should have the 'ignoresdpversion' option set.
1120 * The 'nat' option has now been been changed to have yes, no, force_rport, and
1121 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
1122 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
1123 remote side requests it and disables symmetric RTP support. Setting it to
1124 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
1125 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
1126 and enables symmetric RTP support.
1127 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
1128 response. This permits the master channel to know how each channel dialled
1129 in a multi-channel setup resolved in an individual way. This carries a
1130 performance penalty and can be disabled in sip.conf using the
1131 'storesipcause' option.
1132 * Added 'externtcpport' and 'externtlsport' options to allow custom port
1133 configuration for the externip and externhost options when tcp or tls is used.
1134 * Added support for message body (stored in content variable) to SIP NOTIFY message
1135 accessible via AMI and CLI.
1136 * Added 'media_address' configuration option which can be used to explicitly specify
1137 the IP address to use in the SDP for media (audio, video, and text) streams.
1138 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
1139 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
1141 * Added 'use_q850_reason' configuration option for generating and parsing
1142 if available Reason: Q.850;cause=<cause code> header. It is implemented
1143 in some gateways for better passing PRI/SS7 cause codes via SIP.
1144 * When dialing SIP peers, a new component may be added to the end of the dialstring
1145 to indicate that a specific remote IP address or host should be used when dialing
1146 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
1147 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
1148 ability to selectively force bridged channels to also be encrypted is also
1149 implemented. Branching in the dialplan can be done based on whether or not
1150 a channel has secure media and/or signaling.
1151 * Added directmediapermit/directmediadeny to limit which peers can send direct media
1153 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
1154 Charge messages to snom phones.
1155 * Added support for G.719 media streams.
1156 * Added support for 16khz signed linear media streams.
1157 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
1158 RTP has been outfitted with the same abilities.
1159 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
1160 available in device configurations as well as in the dial plan.
1161 * Addition of the 'subscribe_network_change' option for turning on and off
1162 res_stun_monitor module support in chan_sip.
1163 * Addition of the 'auth_options_requests' option for turning on and off
1164 authentication for OPTIONS requests in chan_sip.
1168 * Add #tryinclude statement for config files. This provides the same
1169 functionality as the #include statement however an asterisk module will
1170 still load if the filename does not exist. Using the #include statement
1171 Asterisk will not allow the module to load.
1175 * Added rtsavesysname option into iax.conf to allow the systname to be saved
1176 on realtime updates.
1177 * Added the ability for chan_iax2 to inform the dialplan whether or not
1178 encryption is being used. This interoperates with the SIP SRTP implementation
1179 so that a secure SIP call can be bridged to a secure IAX call when the
1180 dialplan requires bridged channels to be "secure".
1181 * Addition of the 'subscribe_network_change' option for turning on and off
1182 res_stun_monitor module support in chan_iax.
1187 * Added ability to preset channel variables on indicated lines with the setvar
1188 configuration option. Also, clearvars=all resets the list of variables back
1190 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
1191 See configs/res_pktccops.conf for more information.
1193 XMPP Google Talk/Jingle changes
1194 -------------------------------
1195 * Added the externip option to gtalk.conf.
1196 * Added the stunaddr option to gtalk.conf which allows for the automatic
1197 retrieval of the external ip from a stun server.
1201 * Added 'p' option to PickupChan() to allow for picking up channel by the first
1202 match to a partial channel name.
1203 * Added .m3u support for Mp3Player application.
1204 * Added progress option to the app_dial D() option. When progress DTMF is
1205 present, those values are sent immediately upon receiving a PROGRESS message
1206 regardless if the call has been answered or not.
1207 * Added functionality to the app_dial F() option to continue with execution
1208 at the current location when no parameters are provided.
1209 * Added the 'a' option to app_dial to answer the calling channel before any
1210 announcements or macros are executed.
1211 * Modified app_dial to set answertime when the called channel answers even if
1212 the called channel hangs up during playback of an announcement.
1213 * Modified app_dial 'r' option to support an additional parameter to play an
1214 indication tone from indications.conf
1215 * Added c() option to app_chanspy. This option allows custom DTMF to be set
1216 to cycle through the next available channel. By default this is still '*'.
1217 * Added x() option to app_chanspy. This option allows DTMF to be set to
1218 exit the application.
1219 * The Voicemail application has been improved to automatically ignore messages
1220 that only contain silence.
1221 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
1222 associated mailbox(es) to be greetings-only.
1223 * The ChanSpy application now has the 'S' option, which makes the application
1224 automatically exit once it hits a point where no more channels are available
1226 * The ChanSpy application also now has the 'E' option, which spies on a single
1227 channel and exits when that channel hangs up.
1228 * The MeetMe application now turns on the DENOISE() function by default, for
1229 each participant. In our tests, this has significantly decreased background
1230 noise (especially noisy data centers).
1231 * Voicemail now permits storage of secrets in a separate file, located in the
1232 spool directory of each individual user. The control for this is located in
1233 the "passwordlocation" option in voicemail.conf. Please see the sample
1234 configuration for more information.
1235 * The ChanIsAvail application now exposes the returned cause code using a separate
1236 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
1237 * Added 'd' option to app_followme. This option disables the "Please hold"
1239 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
1240 received will terminate recording.
1241 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
1242 Previously the folder could only be set per context, but has now been extended
1243 using the imapfolder option.
1244 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
1245 * Voicemail now allows the pager date format to be specified separately from the
1247 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
1248 to allow joining, leaving, and sending text to group chats.
1249 * MeetMe has a new option 'G' to play an announcement before joining a conference.
1250 * Page has a new option 'A(x)' which will playback an announcement simultaneously
1251 to all paged phones (and optionally excluding the caller's one using the new
1252 option 'n') before the call is bridged.
1253 * The 'f' option to Dial has been augmented to take an optional argument. If no
1254 argument is provided, the 'f' option works as it always has. If an argument is
1255 provided, then the connected party information of all outgoing channels created
1256 during the Dial will be set to the argument passed to the 'f' option.
1257 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
1259 * The OSP lookup application adds in/outbound network ID, optional security,
1260 number portability, QoS reporting, destination IP port, custom info and service
1262 * Added new application VMSayName that will play the recorded name of the voicemail
1263 user if it exists, otherwise will play the mailbox number.
1264 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
1265 retrieve state for a particular bridge, where <name> is the conference name
1266 * app_directory now allows exiting at any time using the operator or pound key.
1267 * Voicemail now supports setting a locale per-mailbox.
1268 * Two new applications are provided for declining counting phrases in multiple
1269 languages. See the application notes for SayCountedNoun and SayCountedAdj for
1271 * Voicemail now runs the externnotify script when pollmailboxes is activated and
1273 * Voicemail now includes rdnis within msgXXXX.txt file.
1274 * ExternalIVR now supports IPv6 addresses.
1275 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
1276 at https://wiki.asterisk.org/wiki/x/oQBB
1277 * ParkedCall and Park can now specify the parking lot to use.
1281 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
1282 over SRV records associated with a specific service. From the CLI, type
1283 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
1284 details on how these may be used.
1285 * PITCH_SHIFT dialplan function added. This function can be used to modify the
1286 pitch of a channel's tx and rx audio streams.
1287 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
1288 setting various connected line and redirecting party information.
1289 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
1290 support ISDN subaddressing.
1291 * The CHANNEL() function now supports the "name" and "checkhangup" options.
1292 * For DAHDI channels, the CHANNEL() dialplan function now allows
1293 the dialplan to request changes in the configuration of the active
1294 echo canceller on the channel (if any), for the current call only.
1297 exten => s,n,Set(CHANNEL(echocan_mode)=off)
1299 The possible values are:
1301 on - normal mode (the echo canceller is actually reinitialized)
1303 fax - FAX/data mode (NLP disabled if possible, otherwise completely
1305 voice - voice mode (returns from FAX mode, reverting the changes that
1306 were made when FAX mode was requested)
1307 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
1308 and setting variables on the channel which created the current channel.
1309 Administrators should take care to avoid naming conflicts, when multiple
1310 channels are dialled at once, especially when used with the Local channel
1311 construct (which all could set variables on the master channel). Usage
1312 of the HASH() dialplan function, with the key set to the name of the slave
1313 channel, is one approach that will avoid conflicts.
1314 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
1316 * func_odbc now allows multiple row results to be retrieved without using
1317 mode=multirow. If rowlimit is set, then additional rows may be retrieved
1318 from the same query by using the name of the function which retrieved the
1319 first row as an argument to ODBC_FETCH().
1320 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
1321 dialplan. This function returns the content of the received message.
1322 * Added REPLACE, which searches a given variable name for a set of characters,
1323 then either replaces them with a single character or deletes them.
1324 * Added PASSTHRU, which literally passes the same argument back as its return
1325 value. The intent is to be able to use a literal string argument to
1326 functions that currently require a variable name as an argument.
1327 * HASH-associated variables now can be inherited across channel creation, by
1328 prefixing the name of the hash at assignment with the appropriate number of
1329 underscores, just like variables.
1330 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
1331 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
1332 whether or not channels that are bridged to the current channel will be
1333 required to have secure signaling and/or media.
1334 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
1335 the current channel has secure signaling and/or media.
1336 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
1337 "no_media_path" option.
1338 Returns "0" if there is a B channel associated with the call.
1339 Returns "1" if no B channel is associated with the call. The call is either
1340 on hold or is a call waiting call.
1341 * Added option to dialplan function CDR(), the 'f' option
1342 allows for high resolution times for billsec and duration fields.
1343 * FILE() now supports line-mode and writing.
1344 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
1345 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
1349 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
1350 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
1351 and is set when a dynamic feature is triggered.
1352 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
1353 to dynamically create a new parking lot matching the value this varible is
1355 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
1356 features.conf that should be the base for dynamic parkinglots.
1357 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
1358 parkinglot should have.
1359 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
1360 parkinglot should have.
1361 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
1366 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
1367 timeout has expired.
1368 * Added 'R' option to app_queue. This option stops moh and indicates ringing
1369 to the caller when an Agent's phone is ringing. This can be used to indicate
1370 to the caller that their call is about to be picked up, which is nice when
1371 one has been on hold for an extened period of time.
1372 * A new config option, penaltymemberslimit, has been added to queues.conf.
1373 When set this option will disregard penalty settings when a queue has too
1375 * A new option, 'I' has been added to both app_queue and app_dial.
1376 By setting this option, Asterisk will not update the caller with
1377 connected line changes or redirecting party changes when they occur.
1378 * A 'relative-periodic-announce' option has been added to queues.conf. When
1379 enabled, this option will cause periodic announce times to be calculated
1380 from the end of announcements rather than from the beginning.
1381 * The autopause option in queues.conf can be passed a new value, "all." The
1382 result is that if a member becomes auto-paused, he will be paused in all
1383 queues for which he is a member, not just the queue that failed to reach
1385 * Added dialplan function QUEUE_EXISTS to check if a queue exists
1386 * The queue logger now allows events to optionally propagate to a file,
1387 even when realtime logging is turned on. Additionally, realtime logging
1388 supports sending the event arguments to 5 individual fields, although it
1389 will fallback to the previous data definition, if the new table layout is
1392 mISDN channel driver (chan_misdn) changes
1393 ----------------------------------------
1394 * Added display_connected parameter to misdn.conf to put a display string
1395 in the CONNECT message containing the connected name and/or number if
1396 the presentation setting permits it.
1397 * Added display_setup parameter to misdn.conf to put a display string
1398 in the SETUP message containing the caller name and/or number if the
1399 presentation setting permits it.
1400 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
1401 indicate the dialplan settings are to be obtained from the asterisk
1403 * Made misdn.conf parameter callerid accept the "name" <number> format
1404 used by the rest of the system.
1405 * Made use the nationalprefix and internationalprefix misdn.conf
1406 parameters to prefix any received number from the ISDN link if that
1407 number has the corresponding Type-Of-Number. NOTE: This includes
1408 comparing the incoming call's dialed number against the MSN list.
1409 * Added the following new parameters: unknownprefix, netspecificprefix,
1410 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
1411 received number from the ISDN link if that number has the corresponding
1413 * Added new dialplan application misdn_command which permits controlling
1414 the CCBS/CCNR functionality.
1415 * Added new dialplan function mISDN_CC which permits retrieval of various
1416 values from an active call completion record.
1417 * For PTP, you should manually send the COLR of the redirected-to party
1418 for an incomming redirected call if the incoming call could experience
1419 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
1420 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
1421 if the REDIRECTING(from-num) is not empty.
1422 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
1423 option on all of the REDIRECTING statements before dialing the
1424 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
1425 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
1426 redirecting-to presentation (COLR) when it becomes available.
1427 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
1430 thirdparty mISDN enhancements
1431 -----------------------------
1432 mISDN has been modified by Digium, Inc. to greatly expand facility message
1434 * Enhanced COLP support for call diversion and transfer.
1435 * CCBS/CCNR support.
1437 The latest modified mISDN v1.1.x based version is available at:
1438 http://svn.digium.com/svn/thirdparty/mISDN/trunk
1439 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
1441 Tagged versions of the modified mISDN code are available under:
1442 http://svn.digium.com/svn/thirdparty/mISDN/tags
1443 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
1445 libpri channel driver (chan_dahdi) DAHDI changes
1446 -------------------------------------------
1447 * The channel variable PRIREDIRECTREASON is now just a status variable
1448 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
1449 to read and alter the reason.
1450 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
1451 redirected-to party for an incomming redirected call if the incoming call
1452 could experience further redirects. Just set the
1453 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
1454 to the COLR. A call has been redirected if the REDIRECTING(count) is not
1456 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
1457 use the inhibit(i) option on all of the REDIRECTING statements before
1458 dialing the redirected-to party. You still have to set the
1459 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
1460 will update the redirecting-to presentation (COLR) when it becomes available.
1461 * Added the ability to ignore calls that are not in a Multiple Subscriber
1462 Number (MSN) list for PTMP CPE interfaces.
1463 * Added dynamic range compression support for dahdi channels. It is
1464 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
1465 * Added support for ISDN calling and called subaddress with partial support
1466 for connected line subaddress.
1467 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
1468 * Added handling of received HOLD/RETRIEVE messages and the optional ability
1469 to transfer a held call on disconnect similar to an analog phone.
1470 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
1471 Will reroute/deflect an outgoing call when receive the message.
1472 Can use the DAHDISendCallreroutingFacility to send the message for the
1474 * Added standard location to add options to chan_dahdi dialing:
1475 Dial(DAHDI/g1[/extension[/options]])
1478 R Reverse charging indication
1479 * Added Reverse Charging Indication (Collect calls) send/receive option.
1480 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
1481 Dial(DAHDI/g1/extension/R)
1482 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
1483 (requires latest LibPRI)
1484 * Added ability to send/receive keypad digits in the SETUP message.
1485 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
1486 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
1487 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
1488 (requires latest LibPRI)
1489 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
1490 to eliminate tromboned calls. A tromboned call goes out an interface and comes
1491 back into the same interface. Tromboned calls happen because of call routing,
1492 call deflection, call forwarding, and call transfer.
1493 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
1494 * Added the ability to support call waiting calls. (The SETUP has no B channel
1496 * Added Malicious Call ID (MCID) event to the AMI call event class.
1497 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
1499 Asterisk Manager Interface
1500 --------------------------
1501 * The Hangup action now accepts a Cause header which may be used to
1502 set the channel's hangup cause.
1503 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
1504 to specify a separate .pem file to hold a private key. By default sslcert
1505 is used to hold both the public and private key.
1506 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
1507 for options containing the 'tls' prefix. For example, 'sslenable' is now
1508 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
1509 across all .conf files. All affected sample.conf files have been modified to
1510 reflect this change. Previous options such as 'sslenable' still work,
1511 but options with the 'tls' prefix are preferred.
1512 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
1513 in a channel. (res_mutestream.so)
1514 * The configuration file manager.conf now supports a channelvars option, which
1515 specifies a list of channel variables to include in each channel-oriented
1517 * The redirect command now has new parameters ExtraContext, ExtraExtension,
1518 and ExtraPriority to allow redirecting the second channel to a different
1519 location than the first.
1520 * Added new event "JabberStatus" in the Jabber module to monitor buddies
1522 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
1523 in a MixMonitor recording.
1524 * The 'iax2 show peers' output is now similar to the expected output of
1526 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
1528 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
1529 AOC-E messages on a channel.
1530 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
1531 conform more closely to similar events.
1532 * Added a new eventfilter option per user to allow whitelisting and blacklisting
1534 * Added optional parkinglot variable for park command.
1535 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
1536 if CallerIDNum and CallerIDName headers are also present.
1538 Channel Event Logging
1539 ---------------------
1540 * A new interface, CEL, is introduced here. CEL logs single events, much like
1541 the AMI, but it differs from the AMI in that it logs to db backends much
1542 like CDR does; is based on the event subsystem introduced by Russell, and
1543 can share in all its benefits; allows multiple backends to operate like CDR;
1544 is specialized to event data that would be of concern to billing sytems,
1545 like CDR. Backends for logging and accounting calls have been produced,
1546 but a new CDR backend is still in development.
1550 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
1551 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
1552 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
1553 * Multiple files and formats can now be specified in cdr_custom.conf.
1554 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
1555 See configs/cdr_syslog.conf.sample for more information.
1556 * A 'sequence' field has been added to CDRs which can be combined with
1557 linkedid or uniqueid to uniquely identify a CDR.
1558 * Handling of billsec and duration field has changed. If your table definition
1559 specifies those fields as float,double or similar they will now be logged with
1560 microsecond accuracy instead of a whole integer.
1562 Calendaring for Asterisk
1563 ------------------------
1564 * A new set of modules were added supporing calendar integration with Asterisk.
1565 Dialplan functions for reading from and writing to calendars are included,
1566 as well as the ability to execute dialplan logic upon calendar event notifications.
1567 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
1568 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
1569 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
1570 2003 support does not support forms-based authentication).
1572 Call Completion Supplementary Services for Asterisk
1573 ---------------------------------------------------
1574 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
1575 DAHDI/ISDN supports call completion for the following switch types:
1576 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
1577 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
1579 Multicast RTP Support
1580 ---------------------
1581 * A new RTP engine and channel driver have been added which supports Multicast RTP.
1582 The channel driver can be used with the Page application to perform multicast RTP
1583 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
1584 Type can be either basic or linksys.
1585 Destination is the IP address and port for the RTP packets.
1586 Control address is specific to the linksys type and is used for sending the control
1587 packets unique to them.
1589 Security Events Framework
1590 -------------------------
1591 * Asterisk has a new C API for reporting security events. The module res_security_log
1592 sends these events to the "security" logger level. Currently, AMI is the only
1593 Asterisk component that reports security events. However, SIP support will be
1594 coming soon. For more information on the security events framework, see the
1595 "Asterisk Security Framework" section of the Asterisk wiki at
1596 https://wiki.asterisk.org/wiki/x/wgBQ
1597 * SIP support was added in Asterisk 10
1598 * This API now supports IPv6 addresses
1602 * A technology independent fax frontend (res_fax) has been added to Asterisk.
1603 * A spandsp based fax backend (res_fax_spandsp) has been added.
1604 * The app_fax module has been deprecated in favor of the res_fax module and
1605 the new res_fax_spandsp backend.
1606 * The SendFAX and ReceiveFAX applications now send their log messages to a
1607 'fax' logger level, instead of to the generic logger levels. To see these
1608 messages, the system's logger.conf file will need to direct the 'fax' logger
1609 level to one or more destinations; the logger.conf.sample file includes an
1610 example of how to do this. Note that if the 'fax' logger level is *not*
1611 directed to at least one destination, log messages generated by these
1612 applications will be lost, and that if the 'fax' logger level is directed to
1613 the console, the 'core set verbose' and 'core set debug' CLI commands will
1614 have no effect on whether the messages appear on the console or not.
1618 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
1619 Now, in order to enable transmitting silence during record the transmit_silence
1620 option should be used. transmit_silence_during_record remains a valid option, but
1621 defaults to the behavior of the transmit_silence option.
1622 * Addition of the Unit Test Framework API for managing registration and execution
1623 of unit tests with the purpose of verifying the operation of C functions.
1624 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
1625 XMPP text messages to the remote JID.
1626 * Modules.conf has a new option - "require" - that marks a module as critical for
1627 the execution of Asterisk.
1628 If one of the required modules fail to load, Asterisk will exit with a return
1630 * An 'X' option has been added to the asterisk application which enables #exec support.
1631 This allows #exec to be used in asterisk.conf.
1632 * jabber.conf supports a new option auth_policy that toggles auto user registration.
1633 * A new lockconfdir option has been added to asterisk.conf to protect the
1634 configuration directory (/etc/asterisk by default) during reloads.
1635 * The parkeddynamic option has been added to features.conf to enable the creation
1636 of dynamic parkinglots.
1637 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
1638 the reportalarms config option.
1639 * chan_dahdi supports dialing configuring and dialing by device file name.
1640 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
1641 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
1642 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
1643 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
1644 Handy for the above name-based syntax as it does not depend on
1645 initialization order.
1646 * The Realtime dialplan switch now caches entries for 1 second. This provides a
1647 significant increase in performance (about 3X) for installations using this switchtype.
1648 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
1649 AIS. For more information, please see the Distributed Device State section of the
1650 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1651 * The addition of G.719 pass-through support.
1652 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
1653 during device configuration.
1654 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
1655 have less than 3 lines on the LCD.
1656 * Realtime now supports database failover. See the sample extconfig.conf for details.
1657 * The addition of improved translation path building for wideband codecs. Sample
1658 rate changes during translation are now avoided unless absolutely necessary.
1659 * The addition of the res_stun_monitor module for monitoring and reacting to network
1660 changes while behind a NAT.
1661 * DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf.
1662 DTMF Valid/Invalid number of hits/misses can be set in dsp.conf.
1663 These allow support for any Administration. Default is AT&T values.
1667 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
1668 optionally accept a filename, to apply the setting only to the code generated from
1669 that source file when Asterisk was built. However, there are some modules in Asterisk
1670 that are composed of multiple source files, so this did not result in the behavior
1671 that users expected. In this version, 'core set debug' and 'core set verbose'
1672 can optionally accept *module* names instead (with or without the .so extension),
1673 which applies the setting to the entire module specified, regardless of which source
1674 files it was built from.
1675 * New 'manager show settings' command showing the current settings loaded from
1677 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
1678 the channel hangup request to all channels.
1679 * Added a "core reload" CLI command that executes a global reload of Asterisk.
1681 ------------------------------------------------------------------------------
1682 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
1683 ------------------------------------------------------------------------------
1687 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
1688 Snom phones use this for call pickup of extensions that the phone is
1690 * Added support for setting the domain in the URI for caller of an
1691 outbound call by using the SIPFROMDOMAIN channel variable.
1692 * Added a new configuration option "remotesecret" for authentication to
1693 remote services. For backwards compatibility, "secret" still has the
1694 same function as before, but now you can configure both a remote secret and a
1695 local secret for mutual authentication.
1696 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
1697 the sound will be played to the target of an attended transfer
1698 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
1699 finer control over how many peers Asterisk will qualify and the gap between them
1700 when all peers need to be qualified at the same time.
1701 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
1702 (either globally or for a specific peer), chan_sip will treat any SDP data
1703 it receives as new data and update the media stream accordingly. By
1704 default, Asterisk will only modify the media stream if the SDP session
1705 version received is different from the current SDP session version. This
1706 option is required to interoperate with devices that have non-standard SDP
1707 session version implementations (observed with Microsoft OCS). This option
1708 is disabled by default.
1709 * The parsing of register => lines in sip.conf has been modified to allow a port
1710 to be present in the "user" portion. Please see the sip.conf.sample file for more
1712 * Added support for subscribing to MWI on a remote server and making the status available
1713 as a mailbox. Please see the sip.conf.sample file for more information.
1714 * Added a function to remove SIP headers added in the dialplan before the
1715 first INVITE is generated - SIPRemoveHeader()
1716 * Channel variables set with setvar= in a device configuration is now
1717 set both for inbound and outbound calls.
1718 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
1722 * Added immediate option to iax.conf
1723 * Added forceencryption option to iax.conf
1724 * Added Encryption and Trunk status to manager command "iaxpeers"
1728 * The configuration file now holds separate sections for devices and lines.
1729 Please have a look at configs/skinny.conf.sample and change your skinny.conf
1734 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
1735 support for LibOpenR2. http://www.libopenr2.org/
1736 * The UK option waitfordialtone has been added for use with BT analog
1738 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
1739 is used in conjunction with the 'faxdetect' configuration option. When
1740 'faxbuffers' is used and fax tones are detected, the channel will dynamically
1741 switch to the configured faxbuffers policy. For example, to use 6 buffers
1742 and a 'full' buffer policy for a fax transmission, add:
1744 The faxbuffers configuration will be in affect until the call is torn down.
1745 * Added service message support for 4ESS/5ESS switches.
1749 * For DAHDI channels, the CHANNEL() dialplan function now
1750 supports changing the channel's buffer policy (for the current
1751 call only), using this syntax:
1753 exten => s,n,Set(CHANNEL(buffers)=6,full)
1755 This would change the channel to the 'full' buffer policy and
1756 6 (six) buffers. Possible options for this setting are the same
1757 as those in chan_dahdi.conf.
1758 * Added a new dialplan function, CURLOPT, which permits setting various
1759 options that may be useful with the CURL dialplan function, such as
1760 cookies, proxies, connection timeouts, passwords, etc.
1761 * Permit the syntax and synopsis fields of the corresponding dialplan
1762 functions to be individually set from func_odbc.conf.
1763 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
1764 * func_odbc now may specify an insert query to execute, when the write query
1765 affects 0 rows (usually indicating that no such row exists).
1766 * Added a new dialplan function, LISTFILTER, which permits removing elements
1767 from a set list, by name. Uses the same general syntax as the existing CUT
1768 and FIELDQTY dialplan functions, which also manage lists.
1769 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
1770 obtaining realtime data from the dialplan.
1771 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
1772 a subroutine when using the GoSub() and Return() applications.
1773 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
1774 of "core show function AUDIOHOOK_INHERIT" from the CLI
1775 * Added AES_ENCRYPT. For information on its use, please see the output
1776 of "core show function AES_ENCRYPT" from the CLI
1777 * Added AES_DECRYPT. For information on its use, please see the output
1778 of "core show function AES_DECRYPT" from the CLI
1779 * func_odbc now supports database transactions across multiple queries.
1783 * Scheduled meetme conferences may now have their end times extended by
1785 * app_authenticate now gives the ability to select a prompt other than
1787 * app_directory now pays attention to the searchcontexts setting in
1788 voicemail.conf and will look through all contexts, if no context is
1789 specified in the initial argument.
1790 * A new application, Originate, has been introduced, that allows asynchronous
1791 call origination from the dialplan.
1792 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
1793 in addition to the setting in the "general" context.
1794 * Added ConfBridge dialplan application which does conference bridges without
1795 DAHDI. For information on its use, please see the output of
1796 "core show application ConfBridge" from the CLI.
1800 * The Asterisk CLI has a new command, "channel redirect", which is similar in
1801 operation to the AMI Redirect action.
1802 * extensions.conf now allows you to use keyword "same" to define an extension
1803 without actually specifying an extension. It uses exactly the same pattern
1804 as previously used on the last "exten" line. For example:
1805 exten => 123,1,NoOp(something)
1806 same => n,SomethingElse()
1807 * musiconhold.conf classes of type 'files' can now use relative directory paths,
1808 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
1809 * All deprecated CLI commands are removed from the sourcecode. They are now handled
1810 by the new clialiases module. See cli_aliases.conf.sample file.
1811 * Times within timespecs are now accurate down to the minute. This is a change
1812 from historical Asterisk, which only provided timespecs rounded to the nearest
1813 even (read: evenly divisible by 2) minute mark.
1814 * The realtime switch now supports an option flag, 'p', which disables searches for
1816 * In addition to a time range and date range, timespecs now accept a 5th optional
1817 argument, timezone. This allows you to perform time checks on alternate
1818 timezones, especially if those daylight savings time ranges vary from your
1819 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
1821 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
1822 give you the correct output for an asterisk box behind nat. It will give you the
1823 externhost and localnet settings.
1824 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
1825 can connect calls in passthrough mode, as well as record and play back files.
1826 * Successful and unsuccessful call pickup can now be alerted through sounds, by
1827 using pickupsound and pickupfailsound in features.conf.
1828 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
1829 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
1830 instead of the /var/run/asterisk.pid where it used to be. This will make
1831 installs as non-root easier to manage.
1836 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
1837 be written; they will no longer be explicitly written.
1839 Asterisk Manager Interface
1840 --------------------------
1841 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
1842 a non-empty value) in your request. If you do this, any pending AMI events will
1843 *not* be included in the response to your request as they would normally, but
1844 will be left in the event queue for the next request you make to retrieve. For
1845 some applications, this will allow you to guarantee that you will only see
1846 events in responses to 'WaitEvent' actions, and can better know when to expect them.
1847 To know whether the Asterisk server supports this header or not, your client can
1848 inspect the first response back from the server to see if it includes this header:
1850 Pragma: SuppressEvents
1852 If this is included, the server supports event suppression.
1854 * Added 4 new Actions to list skinny device(s) and line(s)
1860 LDAP Schema File Additions
1861 --------------------------
1862 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
1863 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
1865 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
1866 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
1867 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
1868 * Removed redundant IPaddr (there's already IPAddress)
1869 - Gives more configuration Flags for SIP-Users available (tested)
1870 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
1871 without extensibleObject (which really should be the last resort); gives
1872 also additional possibilities for LDAP-filter
1874 ------------------------------------------------------------------------------
1875 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
1876 ------------------------------------------------------------------------------
1878 Device State Handling
1879 ---------------------
1880 * The event infrastructure in Asterisk got another big update to help support
1881 distributed events. It currently supports distributed device state and
1882 distributed Voicemail MWI (Message Waiting Indication). A new module has
1883 been merged, res_ais, which facilitates communicating events between servers.
1884 It uses the SAForum AIS (Service Availability Forum Application Interface
1885 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
1886 a cluster of Asterisk servers, and to share events between them. For more
1887 information on setting this up, refer to the Distributed Device State section
1888 of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1892 * Added a new dialplan function, AST_CONFIG(), which allows you to access
1893 variables from an Asterisk configuration file.
1894 * The JACK_HOOK function now has a c() option to supply a custom client name.
1895 * Added two new dialplan functions from libspeex for audio gain control and
1896 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
1897 rx directions of a channel from the dialplan.
1898 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
1899 based on other parameters. The default is still to search based on the
1900 forwarding station ID. However, there are new options that allow you to search
1901 based on the message desk terminal ID, or the message desk number.
1902 * TIMEOUT() has been modified to be accurate down to the millisecond.
1903 * ENUM*() functions now include the following new options:
1904 - 'u' returns the full URI and does not strip off the URI-scheme.
1905 - 's' triggers ISN specific rewriting
1906 - 'i' looks for branches into an Infrastructure ENUM tree
1907 - 'd' for a direct DNS lookup without any flipping of digits.
1908 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
1909 * CHANNEL() now has options for the maximum, minimum, and standard or normal
1910 deviation of jitter, rtt, and loss for a call using chan_sip.
1912 DAHDI channel driver (chan_dahdi) Changes
1913 ----------------------------------------
1914 * Channels can now be configured using named sections in chan_dahdi.conf, just
1915 like other channel drivers, including the use of templates.
1916 * The default for pridialplan has changed from 'national' to 'unknown'.
1920 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
1921 to something that matches the pattern a hint will be created using the contents
1922 and variables evaluated.
1923 * Dialplan matching has been extended to allow an extension to return to the
1924 PBX core to wait for more digits. This is done by using the new dialplan
1925 application called "Incomplete". This will permit a whole new level of
1926 extension control, by giving the administrator more control over early
1927 matches employing one of the short-circuit pattern match operators. Note
1928 that custom applications can trigger this same behavior by returning the
1929 special value AST_PBX_INCOMPLETE.
1933 * Directory now permits both first and last names to be matched at the same
1934 time. In addition, the number of digits to enter of the name can be set in
1935 the arguments to Directory; previously, you could enter only 3, regardless
1936 of how many names are in your company. For large companies, this should be
1938 * Voicemail now permits a mailbox setting to wrap around from first to last
1939 messages, if the "messagewrap" option is set to a true value.
1940 * Voicemail now permits an external script to be run, for password validation.
1941 The script should output "VALID" or "INVALID" on stdout, depending upon the
1942 wish to validate or invalidate the password given. Arguments are:
1943 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
1945 * Dial has a new option: F(context^extension^pri), which permits a callee to
1946 continue in the dialplan, at the specified label, if the caller hangs up.
1947 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
1948 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
1949 * The Jack application now has a c() option to supply a custom client name.
1950 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
1951 like the pre-existing whisper mode, except that the spy can also talk to the
1952 participant on the bridged channel as well.
1953 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
1954 to be spoken instead of the channel name or number. For more information on the
1955 use of this option, issue the command "core show application ChanSpy" from the
1957 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
1958 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
1959 words, if using the 'd' option, it is not possible to enter a number to append to
1960 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
1961 change to whisper mode, and pressing 6 will change to barge mode.
1962 * ExternalIVR now takes several options that affect the way it performs, as
1963 well as having several new commands. Please see the External IVR page on the Asterisk
1964 wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
1965 * Added ability to communicate over a TCP socket instead of forking a child process for the
1966 ExternalIVR application.
1967 * ChanIsAvail has a new option, 'a', which will return all available channels instead
1968 of just the first one if you give the function more then one channel to check.
1969 * PrivacyManager now takes an option where you can specify a context where the
1970 given number will be matched. This way you have more control over who is allowed
1971 and it stops the people who blindly enter 10 digits.
1972 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
1973 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
1974 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
1975 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
1976 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
1977 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
1978 * The Dial() application no longer copies the language used by the caller to the callee's
1979 channel. If you desire for the caller's channel's language to be used for file playback
1980 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
1981 * SendImage() no longer hangs up the channel on error; instead, it sets the
1982 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
1983 'UNSUPPORTED'. This change makes SendImage() more consistent with other
1985 * Park has a new option, 's', which silences the announcement of the parking space number.
1986 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
1987 invalid input and will be assumed to mean that no timeout is desired.
1991 * Added DNS manager support to registrations for peers referencing peer entries.
1992 DNS manager runs in the background which allows DNS lookups to be run asynchronously
1993 as well as periodically updating the IP address. These properties allow for
1994 better performance as well as recovery in the event of an IP change.
1995 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
1996 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
1997 These changes also provide performance improvements for call setup and tear down.
1998 * Added ability to specify registration expiry time on a per registration basis in
2000 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
2002 * Added t38pt_usertpsource option. See sip.conf.sample for details.
2003 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
2004 * 'sip show peers' and 'sip show users' display their entries sorted in
2005 alphabetical order, as opposed to the order they were in, in the config
2007 * Videosupport now supports an additional option, "always", which always sets
2008 up video RTP ports, even on clients that don't support it. This helps with
2009 callfiles and certain transfers to ensure that if two video phones are
2010 connected, they will always share video feeds.
2014 * Existing DNS manager lookups extended to check for SRV records.
2015 * IAX2 encryption support has been improved to support periodic key rotation
2016 within a call for enhanced security. The option "keyrotate" has been
2017 provided to disable this functionality to preserve backwards compatibility
2018 with older versions of IAX2 that do not support key rotation.
2022 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
2023 data tree based on the given <path>.
2024 * New CLI command "data show providers" that will display all the registered
2026 * New CLI command, "config reload <file.conf>" which reloads any module that
2027 references that particular configuration file. Also added "config list"
2028 which shows which configuration files are in use.
2029 * New CLI commands, "pri show version" and "ss7 show version" that will
2030 display which version of libpri and libss7 are being used, respectively.
2031 A new API call was added so trunk will now have to be compiled against
2032 a versions of libpri and libss7 that have them or it will not know that
2033 these libraries exist.
2034 * The commands "core show globals", "core set global" and "core set chanvar" has
2035 been deprecated in favor of the more semanticly correct "dialplan show globals",
2036 "dialplan set chanvar" and "dialplan set global".
2037 * New CLI command "dialplan show chanvar" to list all variables associated
2038 with a given channel.
2042 * Addresses managed by DNS manager now can check to see if there is a DNS
2043 SRV record for a given domain and will use that hostname/port if present.
2045 AMI - The manager (TCP/TLS/HTTP)
2046 --------------------------------
2047 * The Status command now takes an optional list of variables to display
2048 along with channel status.
2049 * The QueueEntry event now also includes the channel's uniqueid
2053 * res_odbc no longer has a limit of 1023 total possible unshared connections,
2054 as some people were running into this limit. This limit has been increased
2059 * The TRANSFER queue log entry now includes the the caller's original
2060 position in the transferred-from queue.
2061 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
2062 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
2063 as well as an explanation about timeout options in general
2064 * Added a new option - C - for forcing the "answered elsewhere" flag on
2065 cancellation of calls in to members of the queue. This is to avoid the
2066 call to a member of a queue having the call listed as a "missed call".
2070 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
2071 adaptive capabilities. What this means in practical terms is that if your
2072 realtime table lacks critical fields, Asterisk will now emit warnings to
2073 that effect. Also, some of the realtime drivers have the ability (if
2074 configured) to automatically add those columns to the table with the
2075 correct type and length.
2079 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
2080 the 'setvar' option to cause a given audio file to be played upon completion
2081 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
2082 Skinny channels only.
2083 * You can now compile Asterisk against the Hoard Memory Allocator, see the
2084 Hoard page on the Asterisk wiki for more information:
2085 https://wiki.asterisk.org/wiki/x/pQBB
2086 * Config file variables may now be appended to, by using the '+=' append
2087 operator. This is most helpful when working with long SQL queries in
2088 func_odbc.conf, as the queries no longer need to be specified on a single
2090 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
2091 which will add a second to the billsec when the ending
2092 time is set, if the number in the microseconds field of the end time is
2093 greater than the number of microseconds in the answer time. This allows
2094 users to count the 'initiated' seconds in their billing records.
2096 ------------------------------------------------------------------------------
2097 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
2098 ------------------------------------------------------------------------------
2100 AMI - The manager (TCP/TLS/HTTP)
2101 --------------------------------
2102 * Manager has undergone a lot of changes, all of them documented
2103 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
2104 * Manager version has changed to 1.1
2105 * Added a new action 'CoreShowChannels' to list currently defined channels
2106 and some information about them.
2107 * Added a new action 'SIPshowregistry' to list SIP registrations.
2108 * Added TLS support for the manager interface and HTTP server
2109 * Added the URI redirect option for the built-in HTTP server
2110 * The output of CallerID in Manager events is now more consistent.
2111 CallerIDNum is used for number and CallerIDName for name.
2112 * Enable https support for builtin web server.
2113 See configs/http.conf.sample for details.
2114 * Added a new action, GetConfigJSON, which can return the contents of an
2115 Asterisk configuration file in JSON format. This is intended to help
2116 improve the performance of AJAX applications using the manager interface
2118 * SIP and IAX manager events now use "ChannelType" in all cases where we
2119 indicate channel driver. Previously, we used a mixture of "Channel"
2120 and "ChannelDriver" headers.
2121 * Added a "Bridge" action which allows you to bridge any two channels that
2122 are currently active on the system.
2123 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
2124 the voicemail users setup.
2125 * Added 'DBDel' and 'DBDelTree' manager commands.
2126 * cdr_manager now reports events via the "cdr" level, separating it from
2127 the very verbose "call" level.
2128 * Manager users are now stored in memory. If you change the manager account
2129 list (delete or add accounts) you need to reload manager.
2130 * Added Masquerade manager event for when a masquerade happens between
2132 * Added "manager reload" command for the CLI
2133 * Lots of commands that only provided information are now allowed under the
2134 Reporting privilege, instead of only under Call or System.
2135 * The IAX* commands now require either System or Reporting privilege, to
2136 mirror the privileges of the SIP* commands.
2137 * Added ability to retrieve list of categories in a config file.
2138 * Added ability to retrieve the content of a particular category.
2139 * Added ability to empty a context.
2140 * Created new action to create a new file.
2141 * Updated delete action to allow deletion by line number with respect to category.
2142 * Added new action insert to add new variable to category at specified line.
2143 * Updated action newcat to allow new category to be inserted in file above another
2145 * Added new event "JitterBufStats" in the IAX2 channel
2146 * Originate now requires the Originate privilege and, if you want to call out
2147 to a subshell, it requires the System privilege, as well. This was done to
2148 enhance manager security.
2149 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
2150 * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
2151 or manager show command Atxfer from the CLI
2152 * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
2153 details or manager show command IAXregistry from the CLI
2157 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
2158 state in the dialplan, as well as creating custom device states that are
2159 controllable from the dialplan.
2160 * Extend CALLERID() function with "pres" and "ton" parameters to
2161 fetch string representation of calling number presentation indicator
2162 and numeric representation of type of calling number value.
2163 * MailboxExists converted to dialplan function
2164 * A new option to Dial() for telling IP phones not to count the call
2165 as "missed" when dial times out and cancels.
2166 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
2167 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
2168 held for any given channel. Also, locks are automatically freed when a
2170 * Added HINT() dialplan function that allows retrieving hint information.
2171 Hints are mappings between extensions and devices for the sake of
2172 determining the state of an extension. This function can retrieve the list
2173 of devices or the name associated with a hint.
2174 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
2176 * Added SYSINFO() dialplan function which allows retrieval of system information
2177 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
2178 the existence of a dialplan target.
2179 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
2180 upper and lower case, respectively.
2181 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
2182 ID for the call (not the Asterisk call ID or unique ID), provided that the
2183 channel driver supports this. For SIP, you get the SIP call-ID for the
2184 bridged channel which you can store in the CDR with a custom field.
2188 * Added CLI permissions, config file: cli_permissions.conf
2189 default is to allow all commands for every local user/group.
2190 Also this new feature added three new CLI commands:
2191 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
2192 - cli reload permissions
2193 - cli show permissions
2194 * New CLI command "core show hint" (usage: core show hint <exten>)
2195 * New CLI command "core show settings"
2196 * Added 'core show channels count' CLI command.
2197 * Added the ability to set the core debug and verbose values on a per-file basis.
2198 * Added 'queue pause member' and 'queue unpause member' CLI commands
2199 * Ability to set process limits ("ulimit") without restarting Asterisk
2200 * Enhanced "agi debug" to print the channel name as a prefix to the debug
2201 output to make debugging on busy systems much easier.
2202 * New CLI commands "dialplan set extenpatternmatching true/false"
2203 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
2204 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
2205 listed in the startup_commands section of cli.conf will get executed.
2206 * Added a CLI command, "devstate change", which allows you to set custom device
2207 states from the func_devstate module that provides the DEVICE_STATE() function
2208 and handling of the "Custom:" devices.
2209 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
2210 sorted into the different possible callbacks, with the number of entries
2211 currently scheduled for each. Gives you a feel for how busy the sip channel
2213 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
2214 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
2215 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
2219 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
2220 option is enabled, Asterisk will watch for a CNG tone in the incoming audio
2221 for a received call. If it is detected, the channel will jump to the
2222 'fax' extension in the dialplan.
2223 * The default SIP useragent= identifier now includes the Asterisk version
2224 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
2225 If set, and the incoming request carries authentication info,
2226 the username to match in the users list is taken from the Digest header
2227 rather than from the From: field. This feature is considered experimental.
2228 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
2229 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
2230 * The "localmask" setting was removed in version 1.2 and the reminder about it
2231 being removed is now also removed.
2232 * A new option "busylevel" for setting a level of calls where asterisk reports
2233 a device as busy, to separate it from call-limit. This value is also added
2234 to the SIP_PEER dialplan function.
2235 * A new realtime family called "sipregs" is now supported to store SIP registration
2236 data. If this family is defined, "sippeers" will be used for configuration and
2237 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
2238 registration data, as before.
2239 * The SIPPEER function have new options for port address, call and pickup groups
2240 * Added support for T.140 realtime text in SIP/RTP
2241 * The "checkmwi" option has been removed from sip.conf, as it is no longer
2242 required due to the restructuring of how MWI is handled. See the descriptions
2243 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
2244 for more information.
2245 * Added rtpdest option to CHANNEL() dialplan function.
2246 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
2247 * SIP now adds a header to the CANCEL if the call was answered by another phone
2248 in the same dial command, or if the new c option in dial() is used.
2249 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
2250 states it is not needed. For phones, however, that do require it the "registertrying" option
2251 has been added so it can be enabled.
2252 * A new option called "callcounter" (global/peer/user level) enables call counters needed
2253 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
2254 used to enable this functionality).
2255 * New settings for timer T1 and timer B on a global level or per device. This makes it
2256 possible to force timeout faster on non-responsive SIP servers. These settings are
2257 considered advanced, so don't use them unless you have a problem.
2258 * Added a dial string option to be able to set the To: header in an INVITE to any
2260 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
2261 the qualify frequency.
2262 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
2263 were not properly torn down due to network or endpoint failures during an established
2265 * Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB
2266 and configs/sip.conf.sample for more information on how it is used.
2267 * Added a new configuration option "authfailureevents" that enables manager events when
2268 a peer can't authenticate properly.
2269 * Added DNS manager support to registrations for peers not referencing a peer entry.
2273 * Added the trunkmaxsize configuration option to chan_iax2.
2274 * Added the srvlookup option to iax.conf
2275 * Added support for OSP. The token is set and retrieved through the CHANNEL()
2278 XMPP Google Talk/Jingle changes
2279 -------------------------------
2280 * Added the bindaddr option to gtalk.conf.
2284 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
2285 * Proper codec support in chan_skinny.
2286 * Added settings for IP and Ethernet QoS requests
2290 * Added separate settings for media QoS in mgcp.conf
2292 Console Channel Driver changes
2293 ------------------------------
2294 * Added experimental support for video send & receive to chan_oss.
2295 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
2298 Phone channel changes (chan_phone)
2299 ----------------------------------
2300 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
2302 H.323 channel Changes
2303 ---------------------
2304 * H323 remote hold notification support added (by NOTIFY message
2305 and/or H.450 supplementary service)
2307 Local channel changes
2308 ---------------------
2309 * The device state functionality in the Local channel driver has been updated
2310 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
2311 to just UNKNOWN if the extension exists.
2312 * Added jitterbuffer support for chan_local. This allows you to use the
2313 generic jitterbuffer on incoming calls going to Asterisk applications.
2314 For example, this would allow you to use a jitterbuffer for an incoming
2315 SIP call to Voicemail by putting a Local channel in the middle. This
2316 feature is enabled by using the 'j' option in the Dial string to the Local
2317 channel in conjunction with the existing 'n' option for local channels.
2318 * A 'b' option has been added which causes chan_local to return the actual channel
2319 that is behind it when queried. This is useful for transfer scenarios as the
2320 actual channel will be transferred, not the Local channel.
2322 Agent channel changes
2323 ----------------------
2324 * The ackcall and endcall options are now supplemented with options acceptdtmf
2325 and enddtmf. These allow for the DTMF keypress to be configurable. The options
2326 default to their old hard-coded values ('#' and '*' respectively) so this should
2327 not break any existing agent installations.
2329 DAHDI channel driver (chan_dahdi) Changes
2330 ----------------------------------------
2331 * SS7 support (via libss7 library)
2332 * In India, some carriers transmit CID via dtmf. Some code has been added
2333 that will handle some situations. The cidstart=polarity_IN choice has been added for
2334 those carriers that transmit CID via dtmf after a polarity change.
2335 * CID matching information is now shown when doing 'dialplan show'.
2336 * Added dahdi show version CLI command.
2337 * Added setvar support to chan_dahdi.conf channel entries.
2338 * Added two new options: mwimonitor and mwimonitornotify. These options allow
2339 you to enable MWI monitoring on FXO lines. When the MWI state changes,
2340 the script specified in the mwimonitornotify option is executed. An internal
2341 event indicating the new state of the mailbox is also generated, so that
2342 the normal MWI facilities in Asterisk work as usual.
2343 * Added signalling type 'auto', which attempts to use the same signalling type
2344 for a channel as configured in DAHDI. This is primarily designed for analog
2345 ports, but will also work for digital ports that are configured for FXS or FXO
2346 signalling types. This mode is also the default now, so if your chan_dahdi.conf
2347 does not specify signalling for a channel (which is unlikely as the sample
2348 configuration file has always recommended specifying it for every channel) then
2349 the 'auto' mode will be used for that channel if possible.
2350 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
2351 state for a channel; also ensured that the DNDState Manager event is
2352 emitted no matter how the DND state is set or cleared.
2356 * Added a new channel driver, chan_unistim. See the Asterisk wiki at
2357 https://wiki.asterisk.org/wiki/x/vgsiAQ and configs/unistim.conf.sample
2358 for details. This new channel driver allows you to use Nortel i2002,
2359 i2004, and i2050 phones with Asterisk.
2360 * Added a new channel driver, chan_console, which uses portaudio as a cross
2361 platform audio interface. It was written as a channel driver that would
2362 work with Mac CoreAudio, but portaudio supports a number of other audio
2363 interfaces, as well. Note that this channel driver requires v19 or higher
2364 of portaudio; older versions have a different API.
2368 * Added the ability to specify arguments to the Dial application when using
2369 the DUNDi switch in the dialplan.
2370 * Added the ability to set weights for responses dynamically. This can be
2371 done using a global variable or a dialplan function. Using the SHELL()
2372 function would allow you to have an external script set the weight for
2374 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
2375 functions will allow you to initiate a DUNDi query from the dialplan,
2376 find out how many results there are, and access each one.
2377 * Added the ability to specifiy a port for a dundi peer.
2381 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
2382 functions will allow you to initiate an ENUM lookup from the dialplan,
2383 and Asterisk will cache the results. ENUMRESULT can be used to access
2384 the results without doing multiple DNS queries.
2388 * Added the ability to customize which sound files are used for some of the
2389 prompts within the Voicemail application by changing them in voicemail.conf
2390 * Added the ability for the "voicemail show users" CLI command to show users
2391 configured by the dynamic realtime configuration method.
2392 * MWI (Message Waiting Indication) handling has been significantly
2393 restructured internally to Asterisk. It is now totally event based
2394 instead of polling based. The voicemail application will notify other
2395 modules that have subscribed to MWI events when something in the mailbox
2397 This also means that if any other entity outside of Asterisk is changing
2398 the contents of mailboxes, then the voicemail application still needs to
2399 poll for changes. Examples of situations that would require this option
2400 are web interfaces to voicemail or an email client in the case of using
2401 IMAP storage. So, two new options have been added to voicemail.conf
2402 to account for this: "pollmailboxes" and "pollfreq". See the sample
2403 configuration file for details.
2404 * Added "tw" language support
2405 * Added support for storage of greetings using an IMAP server
2406 * Added ability to customize forward, reverse, stop, and pause keys for message playback
2407 * SMDI is now enabled in voicemail using the smdienable option.
2408 * A "lockmode" option has been added to asterisk.conf to configure the file
2409 locking method used for voicemail, and potentially other things in the
2410 future. The default is the old behavior, lockfile. However, there is a
2411 new method, "flock", that uses a different method for situations where the
2412 lockfile will not work, such as on SMB/CIFS mounts.
2413 * Added the ability to backup deleted messages, to ease recovery in the case
2414 that a user accidentally deletes a message, and discovers that they need it.
2415 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
2416 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
2417 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
2418 voicemail boxes. The SMDI interface can also poll for MWI changes when some
2419 outside entity is modifying the state of the mailbox (such as IMAP storage or
2420 a web interface of some kind).
2421 * Added the support for marking messages as "urgent." There are two methods to accomplish
2422 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
2423 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
2424 the message as urgent after he has recorded a voicemail by following the voice instructions.
2425 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
2430 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
2431 used across multiple queues.
2432 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
2433 setqueueentryvar options for each queue, see queues.conf.sample for details.
2434 * Added keepstats option to queues.conf which will keep queue
2435 statistics during a reload.
2436 * setinterfacevar option in queues.conf also now sets a variable
2437 called MEMBERNAME which contains the member's name.
2438 * Added 'Strategy' field to manager event QueueParams which represents
2439 the queue strategy in use.
2440 * Added option to run macro when a queue member is connected to a caller,
2441 see queues.conf.sample for details.
2442 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
2443 does not count paused queue members as unavailable.
2444 * Added min-announce-frequency option to queues.conf which allows you to control the
2445 minimum amount of time between queue announcements for use when the caller's queue
2446 position changes frequently.
2447 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
2449 * Added ability for non-realtime queues to have realtime members
2450 * Added the "linear" strategy to queues.
2451 * Added the "wrandom" strategy to queues.
2452 * Added new channel variable QUEUE_MIN_PENALTY
2453 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
2454 rules in queuerules.conf. See configs/queuerules.conf.sample for details
2455 * Added a new parameter for member definition, called state_interface. This may be
2456 used so that a member may be called via one interface but have a different interface's
2457 device state reported.
2458 * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
2459 "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
2460 "manager show command QueueReset."
2461 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
2462 specified by the periodic-announce option, then one will be chosen randomly when it is time
2463 to play a periodic announcment
2464 * New configuration options: announce-position now takes two more values in addition to "yes" and
2465 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
2466 announce-position-limit. By setting announce-position to "limit" callers will only have their
2467 position announced if their position is less than what is specified by announce-position-limit.
2468 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
2469 will be told that their are more than announce-position-limit callers waiting.
2470 * Two new queue log events have been added. An ADDMEMBER event will be logged
2471 when a realtime queue member is added and a REMOVEMEMBER event will be logged
2472 when a realtime queue member is removed. Since there is no calling channel associated
2473 with these events, the string "REALTIME" is placed where the channel's unique id
2474 is typically placed.
2475 * The configuration method for the "joinempty" and "leavewhenempty" options has
2476 changed to a comma-separated list of methods of determining member availability
2477 instead of vague terms such as "yes," "loose," "no," and "strict." These old four
2478 values are still accepted for backwards-compatibility, though.
2479 * The average talktime is now calculated on queues. This information is reported via the
2480 CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
2481 and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
2486 * The 'o' option to provide an optimization has been removed and its functionality
2487 has been enabled by default.
2488 * When a conference is created, the UNIQUEID of the channel that caused it to be
2489 created is stored. Then, every channel that joins the conference will have the
2490 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
2491 callers that come and go from long standing conferences.
2492 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
2493 except it does operations on a channel by name, instead of number in a conference.
2494 This is a very useful feature in combination with the 'X' option to ChanSpy.
2495 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
2497 * Added new RealTime functionality to provide support for scheduled conferencing.
2498 This includes optional messages to the caller if they attempt to join before
2499 the schedule start time, or to allow the caller to join the conference early.
2500 Also included is optional support for limiting the number of callers per
2501 RealTime conference.
2502 * Added the S() and L() options to the MeetMe application. These are pretty
2503 much identical to the S() and L() options to Dial(). They let you set
2504 timeouts for the conference, as well as have warning sounds played to
2505 let the caller know how much time is left, and when it is running out.
2506 * Added the ability to do "meetme concise" with the "meetme" CLI command.
2507 This extends the concise capabilities of this CLI command to include
2508 listing all conferences, instead of an addition to the other sub commands
2509 for the "meetme" command.
2510 * Added the ability to specify the music on hold class used to play into the
2511 conference when there is only one member and the M option is used.
2512 * Added MEETME_INFO dialplan function which provides a way to query
2513 various properties of a Meetme conference.
2514 * Added new admin features: *81: Roll call, *82: eject all, *83: mute all,
2515 and *84: record in-conf
2517 Other Dialplan Application Changes
2518 ----------------------------------
2519 * Argument support for Gosub application
2520 * From the to-do lists: straighten out the app timeout args:
2521 Wait() app now really does 0.3 seconds- was truncating arg to an int.
2522 WaitExten() same as Wait().
2523 Congestion() - Now takes floating pt. argument.
2524 Busy() - now takes floating pt. argument.
2525 Read() - timeout now can be floating pt.
2526 WaitForRing() now takes floating pt timeout arg.
2527 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
2528 * Added 's' option to Page application.
2529 * Added an optional timeout argument to the Page application.
2530 * Added 'E', 'V', and 'P' commands to ExternalIVR.
2531 * Added 'o' and 'X' options to Chanspy.
2532 * Added a new dialplan application, Bridge, which allows you to bridge the
2533 calling channel to any other active channel on the system.
2534 * Added the ability to specify a music on hold class to play instead of ringing
2535 for the SLATrunk application.
2536 * The Read application no longer exits the dialplan on error. Instead, it sets
2537 READSTATUS to ERROR, which you can catch and handle separately.
2538 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
2539 of asking for verification of each name, one at a time.
2540 * Privacy() no longer uses privacy.conf, as all options are specifyable as
2541 direct options to the app.
2542 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
2544 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
2545 * The ChannelRedirect application no longer exits the dialplan if the given channel
2546 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
2547 or NOCHANNEL if the given channel was not found.
2548 * The silencethreshold setting that was previously configurable in multiple
2549 applications is now settable globally via dsp.conf.
2551 Music On Hold Changes
2552 ---------------------
2553 * A new option, "digit", has been added for music on hold classes in
2554 musiconhold.conf. If this is set for a music on hold class, a caller
2555 listening to music on hold can press this digit to switch to listening
2556 to this music on hold class.
2557 * Support for realtime music on hold has been added.
2558 * In conjunction with the realtime music on hold, a general section has
2559 been added to musiconhold.conf, its sole variable is cachertclasses. If this
2560 is set, then music on hold classes found in realtime will be cached in memory.
2564 * AEL upgraded to use the Gosub with Arguments instead
2565 of Macro application, to hopefully reduce the problems
2566 seen with the artificially low stack ceiling that
2567 Macro bumps into. Macros can only call other Macros
2568 to a depth of 7. Tests run using gosub, show depths
2569 limited only by virtual memory. A small test demonstrated
2570 recursive call depths of 100,000 without problems.
2571 -- in addition to this, all apps that allowed a macro
2572 to be called, as in Dial, queues, etc, are now allowing
2573 a gosub call in similar fashion.
2574 * AEL now generates LOCAL(argname) declarations when it
2575 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
2576 etc. That makes the arguments local in scope. The user
2577 can define their own local variables in macros, now,
2578 by saying "local myvar=someval;" or using Set() in this
2579 fashion: Set(LOCAL(myvar)=someval); ("local" is now
2581 * utils/conf2ael introduced. Will convert an extensions.conf
2582 file into extensions.ael. Very crude and unfinished, but
2583 will be improved as time goes by. Should be useful for a
2584 first pass at conversion.
2585 * aelparse will now read extensions.conf to see if a referenced
2586 macro or context is there before issueing a warning.
2587 * AEL parser sets a local channel variable ~~EXTEN~~, to
2588 preserve the value of ${EXTEN} thru switch statements.
2589 * New operator in $[...] expressions: the ~~ operator serves
2590 as a concatenation operator. AT THE MOMENT, it is really only
2591 necessary and useful in AEL, especially in if() expressions.
2592 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
2593 any enclosing double-quotes, and evaluate to the value of a
2594 concatenated with the value of b. For example if a is set to
2595 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
2596 evaluate to xyzabc .
2599 Call Features (res_features) Changes
2600 ------------------------------------
2601 * Added the parkedcalltransfers option to features.conf
2602 * Added parkedcallparking option to control one touch parking w/ parking
2604 * Added parkedcallhangup option to control disconnect feature w/ parking
2606 * Added parkedcallrecording option to control one-touch record w/ parking
2608 * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
2609 parkedcalltransfers option support for multiple parking lots.
2610 * Added BRIDGE_FEATURES variable to set available features for a channel
2611 * The built-in method for doing attended transfers has been updated to
2612 include some new options that allow you to have the transferee sent
2613 back to the person that did the transfer if the transfer is not successful.
2614 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
2615 in features.conf.sample.
2616 * Added support for configuring named groups of custom call features in
2617 features.conf. This means that features can be written a single time, and
2618 then mapped into groups of features for different key mappings or easier
2620 * Updated the ParkedCall application to allow you to not specify a parking
2621 extension. If you don't specify a parking space to pick up, it will grab
2622 the first one available.
2623 * Added cli command 'features reload' to reload call features from features.conf
2624 * Moved into core asterisk binary.
2625 * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
2626 * Added the ability for custom parking lots to be configured with their own
2627 parking extension with the parkext option.
2629 Language Support Changes
2630 ------------------------
2631 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
2632 * Added support for the Hungarian language for saying numbers, dates, and times.
2636 * Added SPEECH commands for speech recognition. A complete listing can be found
2638 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
2639 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
2640 does not behave as expected; the native command needs to be used, instead.
2641 * Added the ability to perform SRV lookups on fast AGI calls. To use this
2642 feature, simply use hagi: instead of agi: as the protocol portion
2643 of the URI parameter to the AGI function call in your dial plan. Also note
2644 that specifying a port number in the AGI URI will disable SRV lookups,
2645 even if you use the hagi: protocol.
2646 * No longer support MSG_OOB flag on HANGUP.
2650 * Added rotatestrategy option to logger.conf, along with two new options:
2651 "timestamp" which will use the time to name the logger files instead of
2652 sequence number; and "rotate", which rotates the names of the log files,
2653 similar to the way syslog rotates files.
2654 * Added exec_after_rotate option to logger.conf, which allows a system
2655 command to be run after rotation. This is primarily useful with
2656 rotatestrategy=rotate, to allow a limit on the number of log files kept
2657 and to ensure that the oldest log file gets deleted.
2658 * Added realtime support for the queue log
2662 * The cdr_manager module has a [mappings] feature, like cdr_custom,
2663 to add fields to the manager event from the CDR variables.
2664 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
2665 backend database CDR table. Specifically, additional, non-standard
2666 columns are supported, merely by setting the corresponding CDR variable in
2667 your dialplan. In addition, you may alias any column to another name (for
2668 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
2669 simply "alias src => ANI" in the configuration file). Records may be
2670 posted to more than one backend, simply by specifying multiple categories
2671 in the configuration file. And finally, you may filter which CDRs get
2672 posted to each backend, by specifying a filter (which the record must
2673 match) for the particular category. Filters are additive (meaning all
2674 rules must match to post that CDR).
2675 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
2676 module. Specifically, you may add additional columns into the table and
2677 they will be set, if you set the corresponding CDR variable name. Also,
2678 if you omit columns in your database table, they will be silently skipped
2679 (but a record will still be inserted, based on what columns remain). Note
2680 that the other two features from cdr_adaptive_odbc (alias and filter) are
2681 not currently supported.
2682 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
2683 has been disabled using the NoCDR application.
2685 Miscellaneous New Modules
2686 -------------------------
2687 * Added a new CDR module, cdr_sqlite3_custom.
2688 * Added a new realtime configuration module, res_config_sqlite
2689 * Added a new codec translation module, codec_resample, which re-samples
2690 signed linear audio between 8 kHz and 16 kHz to help support wideband
2692 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
2693 based on configuration templates that use Asterisk dialplan function and
2694 variable substitution. It should be possible to create phone profiles and
2695 templates that work for the majority of phones provisioned over http. It
2696 is currently only intended to provision a single user account per phone.
2697 An example profile and set of templates for Polycom phones is provided.
2698 NOTE: Polycom firmware is not included, but should be placed in
2699 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
2700 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
2701 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
2702 provided; there is a JACK() application, and a JACK_HOOK() function. Both
2703 interfaces create an input and output JACK port. The application makes
2704 these ports the endpoint of the call. The audio coming from the channel
2705 goes out the output port and whatever comes back in on the input port is
2706 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
2707 audiohook on the channel. This lets you run the audio coming from a
2708 channel through JACK, and whatever comes back in is what gets forwarded
2709 on as the channel's audio. This is very useful for building custom
2710 vocoders or doing recording or analysis of the channel's audio in another
2712 * Added a new module, res_config_curl, which permits using a HTTP POST url
2713 to retrieve, create, update, and delete realtime information from a remote
2714 web server. Note that this module requires func_curl.so to be loaded for
2715 backend functionality.
2716 * Added a new module, res_config_ldap, which permits the use of an LDAP
2717 server for realtime data access.
2718 * Added support for writing and running your dialplan in lua using the pbx_lua
2719 module. See configs/extensions.lua.sample for examples of how to do this.
2723 * Ability to use libcap to set high ToS bits when non-root
2724 on Linux. If configure is unable to find libcap then you
2725 can use --with-cap to specify the path.
2726 * Added maxfiles option to options section of asterisk.conf which allows you to specify
2727 what Asterisk should set as the maximum number of open files when it loads.
2728 * Added the jittertargetextra configuration option.
2729 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
2730 configuration files for the IP channel drivers. The new option is "cos".
2731 This information is also documented on the Asterisk wiki at
2732 https://wiki.asterisk.org/wiki/x/EYBG
2733 * When originating a call using AMI or pbx_spool that fails the reason for failure
2734 will now be available in the failed extension using the REASON dialplan variable.
2735 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
2736 It allows you to configure a prefix for auto-monitor recordings.
2737 * A new extension pattern matching algorithm, based on a trie, is introduced
2738 here, that could noticeably speed up mid-sized to large dialplans.
2739 It is NOT used by default, as duplicating the behaviour of the old pattern
2740 matcher is still under development. A config file option, in extensions.conf,
2741 in the [general] section, called "extenpatternmatchingnew", is by default
2742 set to false; setting that to true will force the use of the new algorithm.
2743 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
2744 be used to switch the algorithms at run time.
2745 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
2746 specifying which socket to use to connect to the running Asterisk daemon
2748 * Performance enhancements to the sched facility, which is used in
2749 the channel drivers, etc. Added hashtabs and doubly-linked lists
2750 to speed up deletion; start at the beginning or end of list to
2752 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
2753 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
2754 Added regression tests to the tests/ dir, also.
2755 * Added a refcount trace feature to astobj2 for those trying to balance
2756 object creation, deletion; work, play; space and time. See the
2757 notes in astobj2.h. Also, see utils/refcounter as well, as a
2758 quick way to find unbalanced refcounts in what could be a sea
2759 of objects that were balanced.
2760 * Added logging to 'make update' command. See update.log
2761 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
2762 do not come from the remote party.
2763 * Added the 'n' option to the SpeechBackground application to tell it to not
2764 answer the channel if it has not already been answered.
2765 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
2766 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
2768 * iLBC source code no longer included (see UPGRADE.txt for details)
2769 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
2770 deadlock is detected, a backtrace of the stack which led to the lock calls
2771 will be output to the CLI.
2772 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
2773 the "core show locks" CLI command will give lock information output as well
2774 as a backtrace of the stack which led to the lock calls.
2775 * users.conf now sports an optional alternateexts property, which permits
2776 allocation of additional extensions which will reach the specified user.
2777 * A new option for the configure script, --enable-internal-poll, has been added
2778 for use with systems which may have a buggy implementation of the poll system
2779 call. If you notice odd behavior such as the CLI being unresponsive on remote
2780 consoles, you may want to try using this option. This option is enabled by default
2781 on Darwin systems since it is known that the Darwin poll() implementation has
2785 --------------------
2786 * In addition to timing from DAHDI, there is a new timing module called
2787 res_timing_timerfd. In order to use this, you must be running Linux with
2788 a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
2789 script will be able to tell if you have the requirements. From menuselect, select
2790 res_timing_timerfd from the Resource Modules menu.
projects / asterisk/asterisk.git / blob