1 ==============================================================================
3 === This file documents the new and/or enhanced functionality added in
4 === the Asterisk versions listed below. This file does NOT include
5 === changes in behavior that would not be backwards compatible with
6 === previous versions; for that information see the UPGRADE.txt file
7 === and the other UPGRADE files for older releases.
9 ==============================================================================
11 ------------------------------------------------------------------------------
12 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
13 ------------------------------------------------------------------------------
17 * The Asterisk build system will now build and install a shared library
18 (libasteriskssl.so) used to wrap various initialization and shutdown functions
19 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
20 that Asterisk can ensure that these functions do *not* get called by any
21 modules that are loaded into Asterisk, since they should only be called once
22 in any single process. If desired, this feature can be disabled by supplying
23 the "--disable-asteriskssl" option to the configure script.
25 * A new make target, 'full', has been added to the Makefile. This performs
26 the same compilation actions as make all, but will also scan the entirety of
27 each source file for documentation. This option is needed to generate AMI
28 event documentation. Note that your system must have Python in order for
29 this make target to succeed.
31 * The optimization portion of the build system has been reworked to avoid
32 broken builds on certain architectures. All architecture-specific
33 optimization has been removed in favor of using -march=native to allow gcc
34 to detect the environment in which it is running when possible. This can
35 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
37 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
38 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
40 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
41 previously parsed the header file to obtain the version of Asterisk, you
42 will now have to go through Asterisk to get the version information.
50 * Added 'F()' option. Similar to the dial option, this can be supplied with
51 arguments indicating where the callee should go after the caller is hung up,
52 or without options specified, the priority after the Queue will be used.
57 * Added menu action admin_toggle_mute_participants. This will mute / unmute
58 all non-admin participants on a conference. The confbridge configuration
59 file also allows for the default sounds played to all conference users when
60 this occurs to be overriden using sound_participants_unmuted and
61 sound_participants_muted.
63 * Added menu action participant_count. This will playback the number of
64 current participants in a conference.
66 * Added announcement configuration option to user profile. If set the sound
67 file will be played to the user, and only the user, upon joining the
73 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
74 channels respectively before the callee channels are called.
79 * Added support for IPv6.
81 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
82 external process will cause the current playlist to be cleared, including
83 stopping any audio file that is currently playing. This is useful when you
84 want to interrupt audio playback only when specific DTMF is entered by the
90 * A new option, 'I' has been added to app_followme. By setting this option,
91 Asterisk will not update the caller with connected line changes when they
92 occur. This is similar to app_dial and app_queue.
94 * The 'N' option is now ignored if the call is already answered.
96 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
97 and caller channels respectively before the callee channels are called.
99 * The winning FollowMe outgoing call is now put on hold if the caller put it on
105 * MixMonitor hooks now have IDs associated with them which can be used to
106 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
107 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
108 now accepts that ID as an argument.
110 * Added 'm' option, which stores a copy of the recording as a voicemail in the
115 * The connect action in app_mysql now allows you to specify a port number to
116 connect to. This is useful if you run a MySQL server on a non-standard
121 * Increased the default number of allowed destinations from 5 to 12.
126 * The app_page application now no longer depends on DAHDI or app_meetme. It
127 has been re-architected to use app_confbridge internally.
132 * Added queue options autopausebusy and autopauseunavail for automatically
133 pausing a queue member when their device reports busy or congestion.
135 * The 'ignorebusy' option for queue members has been deprecated in favor of
136 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
137 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
138 per interface basis. Individual ringinuse values can now be set in
139 queues.conf via an argument to member definitions. Lastly, the queue
140 'ringinuse' setting now only determines defaults for the per member
141 'ringinuse' setting and does not override per member settings like it does
144 * Added 'F()' option. Similar to the dial option, this can be supplied with
145 arguments indicating where the callee should go after the caller is hung up,
146 or without options specified, the priority after the Queue will be used.
148 * Added new option log_member_name_as_agent, which will cause the membername to
149 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
150 state_interface has been set.
155 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
156 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
157 changed arguments to SayUnixTime so that every option is truly optional even
158 when using multiple options (so that j option could be used without having to
159 manually specify timezone and format) There are other benefits, e.g., format
160 can now be used without specifying time zone as well.
165 * Addition of the VM_INFO function - see Function changes.
167 * The imapserver, imapport, and imapflags configuration options can now be
168 overriden on a user by user basis.
170 * When voicemail plays a message's envelope with saycid set to yes, when
171 reaching the caller id field it will play a recording of a file with the same
172 base name as the sender's callerid if there is a similarly named file in
173 <astspooldir>/recordings/callerids/
175 * Voicemails now contains a unique message identifier "msg_id", which is stored
176 in the message envelope with the sound files. IMAP backends will now store
177 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
178 backends will store the message identifier in a "msg_id" column. See
179 UPGRADE.txt for more information.
181 * Added VoiceMailPlayMsg application. This application will play a single
182 voicemail message from a mailbox. The result of the application, SUCCESS or
183 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
188 * Hangup handlers can be attached to channels using the CHANNEL() function.
189 Hangup handlers will run when the channel is hung up similar to the h
190 extension. The hangup_handler_push option will push a GoSub compatible
191 location in the dialplan onto the channel's hangup handler stack. The
192 hangup_handler_pop option will remove the last added location, and optionally
193 replace it with a new GoSub compatible location. The hangup_handler_wipe
194 option will remove all locations on the stack, and optionally add a new
197 * The expression parser now recognizes the ABS() absolute value function,
198 which will convert negative floating point values to positive values.
200 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
201 control of faxdetect.
203 * Addition of the VM_INFO function that can be used to retrieve voicemail
204 user information, such as the email address and full name.
205 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
208 * The REDIRECTING function now supports the redirecting original party id
211 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
212 lets you set some of the configuration options from the [general] section
213 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
214 the key sequence used to activate built-in features, such as blindxfer,
215 and automon. See the built-in documentation for details.
217 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
218 instead of simply the uri. This is the format that MessageSend() can use
219 in the from parameter for outgoing SIP messages.
221 * Added the PRESENCE_STATE function. This allows retrieving presence state
222 information from any presence state provider. It also allows setting
223 presence state information from a CustomPresence presence state provider.
224 See AMI/CLI changes for related commands.
232 * Added a manager event "LocalBridge" for local channel call bridges between
233 the two pseudo-channels created.
238 * Added dialtone_detect option for analog ports to disconnect incoming
239 calls when dialtone is detected.
241 * Added option colp_send to send ISDN connected line information. Allowed
242 settings are block, to not send any connected line information; connect, to
243 send connected line information on initial connect; and update, to send
244 information on any update during a call. Default is update.
249 * A new channel driver named chan_motif has been added which provides support for
250 Google Talk and Jingle in a single channel driver. This new channel driver includes
251 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
252 hold, unhold, and ringing notification. It is also compliant with the current Jingle
253 specification, current Google Jingle specification, and the original Google Talk
259 * Added NAT support for RTP. Setting in config is 'nat', which can be set
260 globally and overriden on a peer by peer basis.
262 * Direct media functionality has been added. Options in config are:
263 directmedia (directrtp) and directrtpsetup (earlydirect)
265 * ChannelUpdate events now contain a CallRef header.
270 * Asterisk will no longer substitute CID number for CID name in the display
271 name field if CID number exists without a CID name. This change improves
272 compatibility with certain device features such as Avaya IP500's directory
275 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
276 created using that setting to not be removed during SIP reload.
278 * Added settings recordonfeature and recordofffeature. When receiving an INFO
279 request with a "Record:" header, this will turn the requested feature on/off.
280 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
281 dynamic features must be enabled and configured properly on the requesting
282 channel for this to function properly.
284 * Add support to realtime for the 'callbackextension' option.
286 * When multiple peers exist with the same address, but differing
287 callbackextension options, incoming requests that are matched by address
288 will be matched to the peer with the matching callbackextension if it is
291 * Two new NAT options, auto_force_rport and auto_comedia, have been added
292 which set the force_rport and comedia options automatically if Asterisk
293 detects that an incoming SIP request crossed a NAT after being sent by
296 * NAT settings are now a combinable list of options. The equivalent of the
297 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
299 * Adds an option send_diversion which can be disabled to prevent
300 diversion headers from automatically being added to INVITE requests.
302 * Add support for lightweight NAT keepalive. If enabled a blank packet will
303 be sent to the remote host at a given interval to keep the NAT mapping open.
304 This can be enabled using the keepalive configuration option.
306 * Add option 'tonezone' to specify country code for indications. This option
307 can be set both globally and overridden for specific peers.
309 * The SIP Security Events Framework now supports IPv6.
311 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
312 between multiple user agents. When set, for directmedia reinvites,
313 Asterisk will not send an immediate reinvite on an incoming call leg. This
314 option is useful when peered with another SIP user agent that is known to
315 send immediate direct media reinvites upon call establishment.
317 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
319 * Add options subminexpiry and submaxexpiry to set limits of subscription
320 timer independently from registration timer settings. The setting of the
321 registration timer limits still is done by options minexpiry, maxexpiry
322 and defaultexpiry. For backwards compatibility the setting of minexpiry
323 and maxexpiry also is used to configure the subscription timer limits if
324 subminexpiry and submaxexpiry are not set in sip.conf.
325 * Set registration timer limits to default values when reloading sip
326 configuration and values are not set by configuration.
328 * When a MESSAGE request is received, the address the request was received from
329 is now saved in the SIP_RECVADDR variable.
331 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
332 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
333 the ANI2/OLI information is set on the channel, which can be retrieved using
334 the CALLERID function.
336 * Peers can now be configured to support negotiation of ICE candidates using
337 the setting icesupport. See res_rtp_asterisk changes for more information.
339 * Added support for format attribute negotiation. See the Codecs changes for
345 * Added skinny version 17 protocol support.
350 * Added ability to use multiple lines for a single phone. This allows multiple
351 calls to occur on a single phone, using callwaiting and switching between calls.
353 * Added option 'sharpdial' allowing end dialing by pressing # key
355 * Added option 'interdigit_timer' to control phone dial timeout
357 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
359 * Added global 'debug' option, that enables debug in channel driver
361 * Added ability to translate on-screen menu in multiple languages. Tested on
362 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
363 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
366 * In addition to English added French and Russian languages for on-screen menus
368 * Reworked dialing number input: added dialing by timeout, immediate dial on
369 on dialplan compare, phone number length now not limited by screen size
371 * Added ability to pickup a call using features.conf defined value and
377 * The minimum DTMF duration can now be configured in asterisk.conf
378 as "mindtmfduration". The default value is (as before) set to 80 ms.
379 (previously it was only available in source code)
381 * Named ACLs can now be specified in acl.conf and used in configurations that
382 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
383 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
384 working ACL. In addition, some CLI commands have been added to provide
385 show information and allow for module reloading - see CLI Changes.
387 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
388 items (separated by commas), and items in the rule can be negated by prefixing
389 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
390 longer necessray to control the order that the 'permit' and 'deny' columns are
391 returned from queries.
393 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
394 be used within the dynamic weight attribute when specifying a mapping.
396 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
397 header, instead of putting the user defined event name there. When enabled
398 the UserDefType header is added for user defined events. This feature is
399 enabled with the setting show_user_defined.
401 * Macro has been deprecated in favor of GoSub. For redirecting and connected
402 line purposes use the following variables instead of their macro equivalents:
403 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
404 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
405 cc_callback_macro in channel configurations.
407 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
412 * A new channel variable, AGIEXITONHANGUP, has been added which allows
413 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
414 AGI application would exit immediately after a channel hangup is detected.
416 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
417 are resolved and each address is attempted in turn until one succeeds or
421 AMI (Asterisk Manager Interface)
423 * Originate now generates an error response if the extension given is not found
426 * MixMonitor will now show IDs associated with the mixmonitor upon creating
427 them if the i(variable) option is used. StopMixMonitor will accept
428 MixMonitorID as an option to close specific MixMonitors.
430 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
431 updated to include information about peers configured with
432 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
433 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
434 returned if auto_force_rport is not enabled.
436 * Added SIPpeerstatus manager command which will generate PeerStatus events
437 similar to the existing PeerStatus events found in chan_sip on demand.
439 * Hangup now can take a regular expression as the Channel option. If you want
440 to hangup multiple channels, use /regex/ as the Channel option. Existing
441 behavior to hanging up a single channel is unchanged, but if you pass a regex,
442 the manager will send you a list of channels back that were hung up.
444 * Support for IPv6 addresses has been added.
446 * AMI Events can now be documented in the Asterisk source. Note that AMI event
447 documentation is only generated when Asterisk is compiled using 'make full'.
448 See the CLI section for commands to display AMI event information.
450 * The AMI Hangup event now includes the AccountCode header so you can easily
451 correlate with AMI Newchannel events.
453 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
454 the StateInterface of the queue member.
456 * Added AMI event SessionTimeout in the Call category that is issued when a
457 call is terminated due to either RTP stream inactivity or SIP session timer
460 * CEL events can now contain a user defined header UserDefType. See core
461 changes for more information.
463 * OOH323 ChannelUpdate events now contain a CallRef header.
465 * Added PresenceState command. This command will report the presence state for
466 the given presence provider.
468 * Added Parkinglots command. This will list all parking lots as a series of
469 AMI Parkinglot events.
471 * Added MessageSend command. This behaves in the same manner as the
472 MessageSend application, and is a technolgoy agnostic mechanism to send out
473 of call text messages.
475 * Added "message" class authorization. This grants an account permission to
476 send out of call messages. Write-only.
481 * The "dialplan add include" command has been modified to create context a context
482 if one does not already exist. For instance, "dialplan add include foo into bar"
483 will create context "bar" if it does not already exist.
485 * A "dialplan remove context" command has been added to remove a context from
488 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
489 filenames of all running mixmonitors on a channel.
491 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
492 numeric instead of 0, 1, or 2.
494 * "stun show status" will show a table describing how the STUN client is
497 * "acl show [named acl]" will show information regarding a Named ACL. The
498 acl module can be reloaded with "reload acl".
500 * Added CLI command to display AMI event information - "manager show events",
501 which shows a list of all known and documented AMI events, and "manager show
502 event [event name]", which shows detail information about a specific AMI
505 * The result of the CLI command "queue show" now includes the state interface
506 information of the queue member.
508 * The command "core set verbose" will now set a separate level of logging for
509 each remote console without affecting any other console.
511 * Added command "cdr show pgsql status" to check connection status
513 * "sip show channel" will now display the complete route set.
515 * Added "presencestate list" command. This command will list all custom
516 presence states that have been set by using the PRESENCE_STATE dialplan
519 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
520 command. This changes a custom presence to a new state.
525 * Codec lists may now be modified by the '!' character, to allow succinct
526 specification of a list of codecs allowed and disallowed, without the
527 requirement to use two different keywords. For example, to specify all
528 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
530 * Add support for parsing SDP attributes, generating SDP attributes, and
531 passing it through. This support includes codecs such as H.263, H.264, SILK,
532 and CELT. You are able to set up a call and have attribute information pass.
533 This should help considerably with video calls.
535 * The iLBC codec can now use a system-provided iLBC library if one is installed,
536 just like the GSM codec.
540 * Asterisk version and build information is now logged at the beginning of a
543 * Threads belonging to a particular call are now linked with callids which get
544 added to any log messages produced by those threads. Log messages can now be
545 easily identified as involved with a certain call by looking at their call id.
546 Call ids may also be attached to log messages for just about any case where
547 it can be determined to be related to a particular call.
549 * Each logging destination and console now have an independent notion of the
550 current verbosity level. Logger.conf now allows an optional argument to
551 the 'verbose' specifier, indicating the level of verbosity sent to that
552 particular logging destination. Additionally, remote consoles now each
553 have their own verbosity level. The command 'core set verbose' will now set
554 a separate level for each remote console without affecting any other
560 * Added 'announcement' option which will play at the start of MOH and between
561 songs in modes of MOH that can detect transitions between songs (eg.
567 * New per parking lot options: comebackcontext and comebackdialtime. See
568 configs/features.conf.sample for more details.
570 * Channel variable PARKER is now set when comebacktoorigin is disabled in
573 * Channel variable PARKEDCALL is now set with the name of the parking lot
574 when a timeout occurs.
580 CDR Postgresql Driver
582 * Added command "cdr show pgsql status" to check connection status
585 CDR Adaptive ODBC Driver
587 * Added schema option for databases that support specifying a schema.
595 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
596 CALENDAR_WRITE has completed successfully.
601 * A new option, 'probation' has been added to rtp.conf
602 RTP in strictrtp mode can now require more than 1 packet to exit learning
603 mode with a new source (and by default requires 4). The probation option
604 allows the user to change the required number of packets in sequence to any
605 desired value. Use a value of 1 to essentially restore the old behavior.
606 Also, with strictrtp on, Asterisk will now drop all packets until learning
607 mode has successfully exited. These changes are based on how pjmedia handles
608 media sources and source changes.
610 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
611 enabled or disabled using the icesupport setting. A variety of other
612 settings have been introduced to configure STUN/TURN connections.
617 * A new module, res_corosync, has been introduced. This module uses the
618 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
619 of Asterisk servers to both Message Waiting Indication (MWI) and/or
620 Device State (presence) information. This module is very similar to, and
621 is a replacement for the res_ais module that was in previous releases of
627 * This module adds a cleaned up, drop-in replacement for res_jabber called
628 res_xmpp. This provides the same externally facing functionality but is
629 implemented differently internally. res_jabber has been deprecated in favor
630 of res_xmpp; please see the UPGRADE.txt file for more information.
635 * The safe_asterisk script has been updated to allow several of its parameters
636 to be set from environment variables. This also enables a custom run
637 directory of Asterisk to be specified, instead of defaulting to /tmp.
639 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
640 its value to determine the directory to assume is the top-level directory of
641 the source tree. If the variable is not set, it defaults to the current
642 behavior and uses the current working directory.
645 ------------------------------------------------------------------------------
646 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
647 ------------------------------------------------------------------------------
651 * Asterisk now has protocol independent support for processing text messages
652 outside of a call. Messages are routed through the Asterisk dialplan.
653 SIP MESSAGE and XMPP are currently supported. There are options in
654 jabber.conf and sip.conf to allow enabling these features.
655 -> jabber.conf: see the "sendtodialplan" and "context" options.
656 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
657 and "outofcall_message_context" options.
658 The MESSAGE() dialplan function and MessageSend() application have been
659 added to go along with this functionality. More detailed usage information
660 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
661 * If real-time text support (T.140) is negotiated, it will be preferred for
662 sending text via the SendText application. For example, via SIP, messages
663 that were once sent via the SIP MESSAGE request would be sent via RTP if
664 T.140 text is negotiated for a call.
668 * parkedmusicclass can now be set for non-default parking lots.
670 Asterisk Manager Interface
671 --------------------------
672 * PeerStatus now includes Address and Port.
673 * Added Hold events for when the remote party puts the call on and off hold
674 for chan_dahdi ISDN channels.
675 * Added new action MeetmeListRooms to list active conferences (shows same
676 data as "meetme list" at the CLI).
677 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
678 Description field that is set by 'description' in the channel configuration
680 * Added Uniqueid header to UserEvent.
681 * Added new action FilterAdd to control event filters for the current session.
682 This requires the system permission and uses the same filter syntax as
683 filters that can be defined in manager.conf
684 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
685 versions had some instances of the event converted, but others were left
686 as-is. All Unlink events should now be converted to Bridge events. The AMI
687 protocol version number was incremented to 1.2 as a result of this change.
690 --------------------------
691 * The HTTP Server can bind to IPv6 addresses.
694 --------------------------
695 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
696 with busydetect. usage example: busypattern=200,200,200,600
699 --------------------------
700 * New 'gtalk show settings' command showing the current settings loaded from
702 * The 'logger reload' command now supports an optional argument, specifying an
703 alternate configuration file to use.
704 * 'dialplan add extension' command will now automatically create a context if
705 the specified context does not exist with a message indicated it did so.
706 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
707 Description field which can be populated with 'description' in the channel
708 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
711 --------------------------
712 * The filter option in cdr_adaptive_odbc now supports negating the argument,
713 thus allowing records which do NOT match the specified filter.
714 * Added ability to log CONGESTION calls to CDR
717 --------------------------
718 * Ability to define custom SILK formats in codecs.conf.
719 * Addition of speex32 audio format with translation.
720 * CELT codec pass-through support and ability to define
721 custom CELT formats in codecs.conf.
722 * Ability to read raw signed linear files with sample rates
723 ranging from 8khz - 192khz. The new file extensions introduced
724 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
725 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
726 Skinny, H.323, etc) can still only support the following codecs:
727 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
728 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
729 Video: h261, h263, h263p, h264, mpeg4
734 --------------------------
735 * New highly optimized and customizable ConfBridge application capable of
736 mixing audio at sample rates ranging from 8khz-96khz.
737 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
738 and bridge profiles on a channel.
739 * CONFBRIDGE_INFO dialplan function capable of retrieving information
740 about a conference such as locked status and number of parties, admins,
742 * Addition of video_mode option in confbridge.conf for adding video support
743 into a bridge profile.
744 * Addition of the follow_talker video_mode in confbridge.conf. This video
745 mode dynamically switches the video feed to always display the loudest talker
746 supplying video in the conference.
750 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
751 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
752 variables from asterisk.conf.
756 * Addition of the JITTERBUFFER dialplan function. This function allows
757 for jitterbuffering to occur on the read side of a channel. By using
758 this function conference applications such as ConfBridge and MeetMe can
759 have the rx streams jitterbuffered before conference mixing occurs.
760 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
762 * Added STRREPLACE function. This function let's the user search a variable
763 for a given string to replace with another string as many times as the
764 user specifies or just throughout the whole string.
765 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
766 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
767 * Added extensions to chan_ooh323 in function CHANNEL()
769 libpri channel driver (chan_dahdi) DAHDI changes
770 --------------------------
771 * Added moh_signaling option to specify what to do when the channel's bridged
772 peer puts the ISDN channel on hold.
773 * Added display_send and display_receive options to control how the display ie
774 is handled. To send display text from the dialplan use the SendText()
775 application when the option is enabled.
776 * Added mcid_send option to allow sending a MCID request on a span.
779 --------------------------
780 * Added setvar option to calendar.conf to allow setting channel variables on
781 notification channels.
782 * Added "calendar show types" CLI command to list registered calendar
786 --------------------------
787 * Added two new options, r and t with file name arguments to record
788 single direction (unmixed) audio recording separate from the bidirectional
789 (mixed) recording. The mixed file name argument is optional now as long
790 as at least one recording option is used.
793 --------------------------
794 * Added a new option, l, which will disable local call optimization for
795 channels involved with the FollowMe thread. Use this option to improve
796 compatability for a FollowMe call with certain dialplan apps, options, and
800 --------------------------
801 * Added option "k" that will automatically close the conference when there's
802 only one person left when a user exits the conference.
805 --------------------------
806 * cel_pgsql now supports the 'extra' column for data added using the
807 CELGenUserEvent() application.
810 --------------------------
811 * Support for defining hints has been added to pbx_lua. See the 'hints' table
812 in the sample extensions.lua file for syntax details.
813 * Applications that perform jumps in the dialplan such as Goto will now
814 execute properly. When pbx_lua detects that the context, extension, or
815 priority we are executing on has changed it will immediately return control
816 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
817 the priority after the currently executing priority.
818 * An autoservice is now started by default for pbx_lua channels. It can be
819 stopped and restarted using the autoservice_stop() and autoservice_start()
823 --------------------------
824 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
825 into a FAXStatus event with an 'Operation' header that will be either
826 'send', 'receive', and 'gateway'.
827 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
828 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
829 feature will handle converting a fax call between an audio T.30 fax terminal
830 and an IFP T.38 fax terminal.
834 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
835 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
836 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
840 * Added general option negative_penalty_invalid default off. when set
841 members are seen as invalid/logged out when there penalty is negative.
842 for realtime members when set remove from queue will set penalty to -1.
843 * Added queue option autopausedelay when autopause is enabled it will be
844 delayed for this number of seconds since last successful call if there
845 was no prior call the agent will be autopaused immediately.
846 * Added member option ignorebusy this when set and ringinuse is not
847 will allow per member control of multiple calls as ringinuse does for
849 * Added global option check_state_unknown to enforce checking of device state
850 when the device state is unknown app_queue will see unknown as available.
854 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
856 * Added 'k' option to MeetMe to automatically kill the conference when there's only
857 one participant left (much like a normal call bridge)
858 * Added extra argument to Originate to set timeout.
862 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
863 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
864 utility in the UTILS section of menuselect. If an existing astdb is found and no
865 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
866 convert an existing astdb to the SQLite3 version automatically at runtime.
870 * Modules marked as deprecated are no longer marked as building by default. Enabling
871 these modules is still available via menuselect.
875 * authdebug is now disabled by default. To enable this functionaility again
876 set authdebug = yes in iax.conf.
880 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
881 releases it was disabled.
885 * The PBX core previously made a call with a non-existing extension test for
886 extension s@default and jump there if the extension existed.
887 This was a bad default behaviour and violated the principle of least surprise.
888 It has therefore been changed in this release. It may affect some
889 applications and configurations that rely on this behaviour. Most channel
890 drivers have avoided this for many releases by testing whether the extension
891 called exists before starting the PBX and generating a local error.
892 This behaviour still exists and works as before.
894 Extension "s" is used when no extension is given in a channel driver,
895 like immediate answer in DAHDI or calling to a domain with no user part
898 ------------------------------------------------------------------------------
899 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
900 ------------------------------------------------------------------------------
904 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
905 now defaults to force_rport. It is very important that phones requiring nat=no be
906 specifically set as such instead of relying on the default setting. If at all
907 possible, all devices should have nat settings configured in the general section as
908 opposed to configuring nat per-device.
909 * Added preferred_codec_only option in sip.conf. This feature limits the joint
910 codecs sent in response to an INVITE to the single most preferred codec.
911 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
912 to be used for the outgoing call. It must be one of the codecs configured
914 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
915 to be used for holding a private key. If tlsprivatekey is not specified,
916 tlscertfile is searched for both public and private key.
917 * Added tlsclientmethod option to sip.conf. This allows the protocol for
918 outbound client connections to be specified.
919 * The sendrpid parameter has been expanded to include the options
920 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
921 header to be sent (equivalent to setting sendrpid=yes) and setting
922 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
923 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
924 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
925 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
926 will accept the SDP even if the SDP version number is not properly incremented,
927 but will generate a warning in the log indicating that the SIP peer that sent
928 the SDP should have the 'ignoresdpversion' option set.
929 * The 'nat' option has now been been changed to have yes, no, force_rport, and
930 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
931 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
932 remote side requests it and disables symmetric RTP support. Setting it to
933 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
934 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
935 and enables symmetric RTP support.
936 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
937 response. This permits the master channel to know how each channel dialled
938 in a multi-channel setup resolved in an individual way. This carries a
939 performance penalty and can be disabled in sip.conf using the
940 'storesipcause' option.
941 * Added 'externtcpport' and 'externtlsport' options to allow custom port
942 configuration for the externip and externhost options when tcp or tls is used.
943 * Added support for message body (stored in content variable) to SIP NOTIFY message
944 accessible via AMI and CLI.
945 * Added 'media_address' configuration option which can be used to explicitly specify
946 the IP address to use in the SDP for media (audio, video, and text) streams.
947 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
948 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
950 * Added 'use_q850_reason' configuration option for generating and parsing
951 if available Reason: Q.850;cause=<cause code> header. It is implemented
952 in some gateways for better passing PRI/SS7 cause codes via SIP.
953 * When dialing SIP peers, a new component may be added to the end of the dialstring
954 to indicate that a specific remote IP address or host should be used when dialing
955 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
956 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
957 ability to selectively force bridged channels to also be encrypted is also
958 implemented. Branching in the dialplan can be done based on whether or not
959 a channel has secure media and/or signaling.
960 * Added directmediapermit/directmediadeny to limit which peers can send direct media
962 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
963 Charge messages to snom phones.
964 * Added support for G.719 media streams.
965 * Added support for 16khz signed linear media streams.
966 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
967 RTP has been outfitted with the same abilities.
968 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
969 available in device configurations as well as in the dial plan.
970 * Addition of the 'subscribe_network_change' option for turning on and off
971 res_stun_monitor module support in chan_sip.
972 * Addition of the 'auth_options_requests' option for turning on and off
973 authentication for OPTIONS requests in chan_sip.
977 * Add #tryinclude statement for config files. This provides the same
978 functionality as the #include statement however an asterisk module will
979 still load if the filename does not exist. Using the #include statement
980 Asterisk will not allow the module to load.
984 * Added rtsavesysname option into iax.conf to allow the systname to be saved
986 * Added the ability for chan_iax2 to inform the dialplan whether or not
987 encryption is being used. This interoperates with the SIP SRTP implementation
988 so that a secure SIP call can be bridged to a secure IAX call when the
989 dialplan requires bridged channels to be "secure".
990 * Addition of the 'subscribe_network_change' option for turning on and off
991 res_stun_monitor module support in chan_iax.
996 * Added ability to preset channel variables on indicated lines with the setvar
997 configuration option. Also, clearvars=all resets the list of variables back
999 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
1000 See configs/res_pktccops.conf for more information.
1002 XMPP Google Talk/Jingle changes
1003 -------------------------------
1004 * Added the externip option to gtalk.conf.
1005 * Added the stunaddr option to gtalk.conf which allows for the automatic
1006 retrieval of the external ip from a stun server.
1010 * Added 'p' option to PickupChan() to allow for picking up channel by the first
1011 match to a partial channel name.
1012 * Added .m3u support for Mp3Player application.
1013 * Added progress option to the app_dial D() option. When progress DTMF is
1014 present, those values are sent immediately upon receiving a PROGRESS message
1015 regardless if the call has been answered or not.
1016 * Added functionality to the app_dial F() option to continue with execution
1017 at the current location when no parameters are provided.
1018 * Added the 'a' option to app_dial to answer the calling channel before any
1019 announcements or macros are executed.
1020 * Modified app_dial to set answertime when the called channel answers even if
1021 the called channel hangs up during playback of an announcement.
1022 * Modified app_dial 'r' option to support an additional parameter to play an
1023 indication tone from indications.conf
1024 * Added c() option to app_chanspy. This option allows custom DTMF to be set
1025 to cycle through the next available channel. By default this is still '*'.
1026 * Added x() option to app_chanspy. This option allows DTMF to be set to
1027 exit the application.
1028 * The Voicemail application has been improved to automatically ignore messages
1029 that only contain silence.
1030 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
1031 associated mailbox(es) to be greetings-only.
1032 * The ChanSpy application now has the 'S' option, which makes the application
1033 automatically exit once it hits a point where no more channels are available
1035 * The ChanSpy application also now has the 'E' option, which spies on a single
1036 channel and exits when that channel hangs up.
1037 * The MeetMe application now turns on the DENOISE() function by default, for
1038 each participant. In our tests, this has significantly decreased background
1039 noise (especially noisy data centers).
1040 * Voicemail now permits storage of secrets in a separate file, located in the
1041 spool directory of each individual user. The control for this is located in
1042 the "passwordlocation" option in voicemail.conf. Please see the sample
1043 configuration for more information.
1044 * The ChanIsAvail application now exposes the returned cause code using a separate
1045 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
1046 * Added 'd' option to app_followme. This option disables the "Please hold"
1048 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
1049 received will terminate recording.
1050 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
1051 Previously the folder could only be set per context, but has now been extended
1052 using the imapfolder option.
1053 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
1054 * Voicemail now allows the pager date format to be specified separately from the
1056 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
1057 to allow joining, leaving, and sending text to group chats.
1058 * MeetMe has a new option 'G' to play an announcement before joining a conference.
1059 * Page has a new option 'A(x)' which will playback an announcement simultaneously
1060 to all paged phones (and optionally excluding the caller's one using the new
1061 option 'n') before the call is bridged.
1062 * The 'f' option to Dial has been augmented to take an optional argument. If no
1063 argument is provided, the 'f' option works as it always has. If an argument is
1064 provided, then the connected party information of all outgoing channels created
1065 during the Dial will be set to the argument passed to the 'f' option.
1066 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
1068 * The OSP lookup application adds in/outbound network ID, optional security,
1069 number portability, QoS reporting, destination IP port, custom info and service
1071 * Added new application VMSayName that will play the recorded name of the voicemail
1072 user if it exists, otherwise will play the mailbox number.
1073 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
1074 retrieve state for a particular bridge, where <name> is the conference name
1075 * app_directory now allows exiting at any time using the operator or pound key.
1076 * Voicemail now supports setting a locale per-mailbox.
1077 * Two new applications are provided for declining counting phrases in multiple
1078 languages. See the application notes for SayCountedNoun and SayCountedAdj for
1080 * Voicemail now runs the externnotify script when pollmailboxes is activated and
1082 * Voicemail now includes rdnis within msgXXXX.txt file.
1083 * ExternalIVR now supports IPv6 addresses.
1084 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
1085 at https://wiki.asterisk.org/wiki/x/oQBB
1086 * ParkedCall and Park can now specify the parking lot to use.
1090 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
1091 over SRV records associated with a specific service. From the CLI, type
1092 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
1093 details on how these may be used.
1094 * PITCH_SHIFT dialplan function added. This function can be used to modify the
1095 pitch of a channel's tx and rx audio streams.
1096 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
1097 setting various connected line and redirecting party information.
1098 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
1099 support ISDN subaddressing.
1100 * The CHANNEL() function now supports the "name" and "checkhangup" options.
1101 * For DAHDI channels, the CHANNEL() dialplan function now allows
1102 the dialplan to request changes in the configuration of the active
1103 echo canceller on the channel (if any), for the current call only.
1106 exten => s,n,Set(CHANNEL(echocan_mode)=off)
1108 The possible values are:
1110 on - normal mode (the echo canceller is actually reinitialized)
1112 fax - FAX/data mode (NLP disabled if possible, otherwise completely
1114 voice - voice mode (returns from FAX mode, reverting the changes that
1115 were made when FAX mode was requested)
1116 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
1117 and setting variables on the channel which created the current channel.
1118 Administrators should take care to avoid naming conflicts, when multiple
1119 channels are dialled at once, especially when used with the Local channel
1120 construct (which all could set variables on the master channel). Usage
1121 of the HASH() dialplan function, with the key set to the name of the slave
1122 channel, is one approach that will avoid conflicts.
1123 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
1125 * func_odbc now allows multiple row results to be retrieved without using
1126 mode=multirow. If rowlimit is set, then additional rows may be retrieved
1127 from the same query by using the name of the function which retrieved the
1128 first row as an argument to ODBC_FETCH().
1129 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
1130 dialplan. This function returns the content of the received message.
1131 * Added REPLACE, which searches a given variable name for a set of characters,
1132 then either replaces them with a single character or deletes them.
1133 * Added PASSTHRU, which literally passes the same argument back as its return
1134 value. The intent is to be able to use a literal string argument to
1135 functions that currently require a variable name as an argument.
1136 * HASH-associated variables now can be inherited across channel creation, by
1137 prefixing the name of the hash at assignment with the appropriate number of
1138 underscores, just like variables.
1139 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
1140 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
1141 whether or not channels that are bridged to the current channel will be
1142 required to have secure signaling and/or media.
1143 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
1144 the current channel has secure signaling and/or media.
1145 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
1146 "no_media_path" option.
1147 Returns "0" if there is a B channel associated with the call.
1148 Returns "1" if no B channel is associated with the call. The call is either
1149 on hold or is a call waiting call.
1150 * Added option to dialplan function CDR(), the 'f' option
1151 allows for high resolution times for billsec and duration fields.
1152 * FILE() now supports line-mode and writing.
1153 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
1154 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
1158 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
1159 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
1160 and is set when a dynamic feature is triggered.
1161 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
1162 to dynamically create a new parking lot matching the value this varible is
1164 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
1165 features.conf that should be the base for dynamic parkinglots.
1166 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
1167 parkinglot should have.
1168 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
1169 parkinglot should have.
1170 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
1175 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
1176 timeout has expired.
1177 * Added 'R' option to app_queue. This option stops moh and indicates ringing
1178 to the caller when an Agent's phone is ringing. This can be used to indicate
1179 to the caller that their call is about to be picked up, which is nice when
1180 one has been on hold for an extened period of time.
1181 * A new config option, penaltymemberslimit, has been added to queues.conf.
1182 When set this option will disregard penalty settings when a queue has too
1184 * A new option, 'I' has been added to both app_queue and app_dial.
1185 By setting this option, Asterisk will not update the caller with
1186 connected line changes or redirecting party changes when they occur.
1187 * A 'relative-periodic-announce' option has been added to queues.conf. When
1188 enabled, this option will cause periodic announce times to be calculated
1189 from the end of announcements rather than from the beginning.
1190 * The autopause option in queues.conf can be passed a new value, "all." The
1191 result is that if a member becomes auto-paused, he will be paused in all
1192 queues for which he is a member, not just the queue that failed to reach
1194 * Added dialplan function QUEUE_EXISTS to check if a queue exists
1195 * The queue logger now allows events to optionally propagate to a file,
1196 even when realtime logging is turned on. Additionally, realtime logging
1197 supports sending the event arguments to 5 individual fields, although it
1198 will fallback to the previous data definition, if the new table layout is
1201 mISDN channel driver (chan_misdn) changes
1202 ----------------------------------------
1203 * Added display_connected parameter to misdn.conf to put a display string
1204 in the CONNECT message containing the connected name and/or number if
1205 the presentation setting permits it.
1206 * Added display_setup parameter to misdn.conf to put a display string
1207 in the SETUP message containing the caller name and/or number if the
1208 presentation setting permits it.
1209 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
1210 indicate the dialplan settings are to be obtained from the asterisk
1212 * Made misdn.conf parameter callerid accept the "name" <number> format
1213 used by the rest of the system.
1214 * Made use the nationalprefix and internationalprefix misdn.conf
1215 parameters to prefix any received number from the ISDN link if that
1216 number has the corresponding Type-Of-Number. NOTE: This includes
1217 comparing the incoming call's dialed number against the MSN list.
1218 * Added the following new parameters: unknownprefix, netspecificprefix,
1219 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
1220 received number from the ISDN link if that number has the corresponding
1222 * Added new dialplan application misdn_command which permits controlling
1223 the CCBS/CCNR functionality.
1224 * Added new dialplan function mISDN_CC which permits retrieval of various
1225 values from an active call completion record.
1226 * For PTP, you should manually send the COLR of the redirected-to party
1227 for an incomming redirected call if the incoming call could experience
1228 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
1229 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
1230 if the REDIRECTING(from-num) is not empty.
1231 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
1232 option on all of the REDIRECTING statements before dialing the
1233 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
1234 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
1235 redirecting-to presentation (COLR) when it becomes available.
1236 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
1239 thirdparty mISDN enhancements
1240 -----------------------------
1241 mISDN has been modified by Digium, Inc. to greatly expand facility message
1243 * Enhanced COLP support for call diversion and transfer.
1244 * CCBS/CCNR support.
1246 The latest modified mISDN v1.1.x based version is available at:
1247 http://svn.digium.com/svn/thirdparty/mISDN/trunk
1248 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
1250 Tagged versions of the modified mISDN code are available under:
1251 http://svn.digium.com/svn/thirdparty/mISDN/tags
1252 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
1254 libpri channel driver (chan_dahdi) DAHDI changes
1255 -------------------------------------------
1256 * The channel variable PRIREDIRECTREASON is now just a status variable
1257 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
1258 to read and alter the reason.
1259 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
1260 redirected-to party for an incomming redirected call if the incoming call
1261 could experience further redirects. Just set the
1262 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
1263 to the COLR. A call has been redirected if the REDIRECTING(count) is not
1265 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
1266 use the inhibit(i) option on all of the REDIRECTING statements before
1267 dialing the redirected-to party. You still have to set the
1268 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
1269 will update the redirecting-to presentation (COLR) when it becomes available.
1270 * Added the ability to ignore calls that are not in a Multiple Subscriber
1271 Number (MSN) list for PTMP CPE interfaces.
1272 * Added dynamic range compression support for dahdi channels. It is
1273 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
1274 * Added support for ISDN calling and called subaddress with partial support
1275 for connected line subaddress.
1276 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
1277 * Added handling of received HOLD/RETRIEVE messages and the optional ability
1278 to transfer a held call on disconnect similar to an analog phone.
1279 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
1280 Will reroute/deflect an outgoing call when receive the message.
1281 Can use the DAHDISendCallreroutingFacility to send the message for the
1283 * Added standard location to add options to chan_dahdi dialing:
1284 Dial(DAHDI/g1[/extension[/options]])
1287 R Reverse charging indication
1288 * Added Reverse Charging Indication (Collect calls) send/receive option.
1289 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
1290 Dial(DAHDI/g1/extension/R)
1291 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
1292 (requires latest LibPRI)
1293 * Added ability to send/receive keypad digits in the SETUP message.
1294 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
1295 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
1296 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
1297 (requires latest LibPRI)
1298 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
1299 to eliminate tromboned calls. A tromboned call goes out an interface and comes
1300 back into the same interface. Tromboned calls happen because of call routing,
1301 call deflection, call forwarding, and call transfer.
1302 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
1303 * Added the ability to support call waiting calls. (The SETUP has no B channel
1305 * Added Malicious Call ID (MCID) event to the AMI call event class.
1306 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
1308 Asterisk Manager Interface
1309 --------------------------
1310 * The Hangup action now accepts a Cause header which may be used to
1311 set the channel's hangup cause.
1312 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
1313 to specify a separate .pem file to hold a private key. By default sslcert
1314 is used to hold both the public and private key.
1315 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
1316 for options containing the 'tls' prefix. For example, 'sslenable' is now
1317 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
1318 across all .conf files. All affected sample.conf files have been modified to
1319 reflect this change. Previous options such as 'sslenable' still work,
1320 but options with the 'tls' prefix are preferred.
1321 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
1322 in a channel. (res_mutestream.so)
1323 * The configuration file manager.conf now supports a channelvars option, which
1324 specifies a list of channel variables to include in each channel-oriented
1326 * The redirect command now has new parameters ExtraContext, ExtraExtension,
1327 and ExtraPriority to allow redirecting the second channel to a different
1328 location than the first.
1329 * Added new event "JabberStatus" in the Jabber module to monitor buddies
1331 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
1332 in a MixMonitor recording.
1333 * The 'iax2 show peers' output is now similar to the expected output of
1335 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
1337 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
1338 AOC-E messages on a channel.
1339 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
1340 conform more closely to similar events.
1341 * Added a new eventfilter option per user to allow whitelisting and blacklisting
1343 * Added optional parkinglot variable for park command.
1344 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
1345 if CallerIDNum and CallerIDName headers are also present.
1347 Channel Event Logging
1348 ---------------------
1349 * A new interface, CEL, is introduced here. CEL logs single events, much like
1350 the AMI, but it differs from the AMI in that it logs to db backends much
1351 like CDR does; is based on the event subsystem introduced by Russell, and
1352 can share in all its benefits; allows multiple backends to operate like CDR;
1353 is specialized to event data that would be of concern to billing sytems,
1354 like CDR. Backends for logging and accounting calls have been produced,
1355 but a new CDR backend is still in development.
1359 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
1360 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
1361 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
1362 * Multiple files and formats can now be specified in cdr_custom.conf.
1363 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
1364 See configs/cdr_syslog.conf.sample for more information.
1365 * A 'sequence' field has been added to CDRs which can be combined with
1366 linkedid or uniqueid to uniquely identify a CDR.
1367 * Handling of billsec and duration field has changed. If your table definition
1368 specifies those fields as float,double or similar they will now be logged with
1369 microsecond accuracy instead of a whole integer.
1371 Calendaring for Asterisk
1372 ------------------------
1373 * A new set of modules were added supporing calendar integration with Asterisk.
1374 Dialplan functions for reading from and writing to calendars are included,
1375 as well as the ability to execute dialplan logic upon calendar event notifications.
1376 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
1377 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
1378 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
1379 2003 support does not support forms-based authentication).
1381 Call Completion Supplementary Services for Asterisk
1382 ---------------------------------------------------
1383 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
1384 DAHDI/ISDN supports call completion for the following switch types:
1385 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
1386 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
1388 Multicast RTP Support
1389 ---------------------
1390 * A new RTP engine and channel driver have been added which supports Multicast RTP.
1391 The channel driver can be used with the Page application to perform multicast RTP
1392 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
1393 Type can be either basic or linksys.
1394 Destination is the IP address and port for the RTP packets.
1395 Control address is specific to the linksys type and is used for sending the control
1396 packets unique to them.
1398 Security Events Framework
1399 -------------------------
1400 * Asterisk has a new C API for reporting security events. The module res_security_log
1401 sends these events to the "security" logger level. Currently, AMI is the only
1402 Asterisk component that reports security events. However, SIP support will be
1403 coming soon. For more information on the security events framework, see the
1404 "Asterisk Security Framework" section of the Asterisk wiki at
1405 https://wiki.asterisk.org/wiki/x/wgBQ
1406 * SIP support was added in Asterisk 10
1407 * This API now supports IPv6 addresses
1411 * A technology independent fax frontend (res_fax) has been added to Asterisk.
1412 * A spandsp based fax backend (res_fax_spandsp) has been added.
1413 * The app_fax module has been deprecated in favor of the res_fax module and
1414 the new res_fax_spandsp backend.
1415 * The SendFAX and ReceiveFAX applications now send their log messages to a
1416 'fax' logger level, instead of to the generic logger levels. To see these
1417 messages, the system's logger.conf file will need to direct the 'fax' logger
1418 level to one or more destinations; the logger.conf.sample file includes an
1419 example of how to do this. Note that if the 'fax' logger level is *not*
1420 directed to at least one destination, log messages generated by these
1421 applications will be lost, and that if the 'fax' logger level is directed to
1422 the console, the 'core set verbose' and 'core set debug' CLI commands will
1423 have no effect on whether the messages appear on the console or not.
1427 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
1428 Now, in order to enable transmitting silence during record the transmit_silence
1429 option should be used. transmit_silence_during_record remains a valid option, but
1430 defaults to the behavior of the transmit_silence option.
1431 * Addition of the Unit Test Framework API for managing registration and execution
1432 of unit tests with the purpose of verifying the operation of C functions.
1433 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
1434 XMPP text messages to the remote JID.
1435 * Modules.conf has a new option - "require" - that marks a module as critical for
1436 the execution of Asterisk.
1437 If one of the required modules fail to load, Asterisk will exit with a return
1439 * An 'X' option has been added to the asterisk application which enables #exec support.
1440 This allows #exec to be used in asterisk.conf.
1441 * jabber.conf supports a new option auth_policy that toggles auto user registration.
1442 * A new lockconfdir option has been added to asterisk.conf to protect the
1443 configuration directory (/etc/asterisk by default) during reloads.
1444 * The parkeddynamic option has been added to features.conf to enable the creation
1445 of dynamic parkinglots.
1446 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
1447 the reportalarms config option.
1448 * chan_dahdi supports dialing configuring and dialing by device file name.
1449 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
1450 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
1451 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
1452 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
1453 Handy for the above name-based syntax as it does not depend on
1454 initialization order.
1455 * The Realtime dialplan switch now caches entries for 1 second. This provides a
1456 significant increase in performance (about 3X) for installations using this switchtype.
1457 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
1458 AIS. For more information, please see the Distributed Device State section of the
1459 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1460 * The addition of G.719 pass-through support.
1461 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
1462 during device configuration.
1463 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
1464 have less than 3 lines on the LCD.
1465 * Realtime now supports database failover. See the sample extconfig.conf for details.
1466 * The addition of improved translation path building for wideband codecs. Sample
1467 rate changes during translation are now avoided unless absolutely necessary.
1468 * The addition of the res_stun_monitor module for monitoring and reacting to network
1469 changes while behind a NAT.
1473 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
1474 optionally accept a filename, to apply the setting only to the code generated from
1475 that source file when Asterisk was built. However, there are some modules in Asterisk
1476 that are composed of multiple source files, so this did not result in the behavior
1477 that users expected. In this version, 'core set debug' and 'core set verbose'
1478 can optionally accept *module* names instead (with or without the .so extension),
1479 which applies the setting to the entire module specified, regardless of which source
1480 files it was built from.
1481 * New 'manager show settings' command showing the current settings loaded from
1483 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
1484 the channel hangup request to all channels.
1485 * Added a "core reload" CLI command that executes a global reload of Asterisk.
1487 ------------------------------------------------------------------------------
1488 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
1489 ------------------------------------------------------------------------------
1493 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
1494 Snom phones use this for call pickup of extensions that the phone is
1496 * Added support for setting the domain in the URI for caller of an
1497 outbound call by using the SIPFROMDOMAIN channel variable.
1498 * Added a new configuration option "remotesecret" for authentication to
1499 remote services. For backwards compatibility, "secret" still has the
1500 same function as before, but now you can configure both a remote secret and a
1501 local secret for mutual authentication.
1502 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
1503 the sound will be played to the target of an attended transfer
1504 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
1505 finer control over how many peers Asterisk will qualify and the gap between them
1506 when all peers need to be qualified at the same time.
1507 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
1508 (either globally or for a specific peer), chan_sip will treat any SDP data
1509 it receives as new data and update the media stream accordingly. By
1510 default, Asterisk will only modify the media stream if the SDP session
1511 version received is different from the current SDP session version. This
1512 option is required to interoperate with devices that have non-standard SDP
1513 session version implementations (observed with Microsoft OCS). This option
1514 is disabled by default.
1515 * The parsing of register => lines in sip.conf has been modified to allow a port
1516 to be present in the "user" portion. Please see the sip.conf.sample file for more
1518 * Added support for subscribing to MWI on a remote server and making the status available
1519 as a mailbox. Please see the sip.conf.sample file for more information.
1520 * Added a function to remove SIP headers added in the dialplan before the
1521 first INVITE is generated - SIPRemoveHeader()
1522 * Channel variables set with setvar= in a device configuration is now
1523 set both for inbound and outbound calls.
1524 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
1528 * Added immediate option to iax.conf
1529 * Added forceencryption option to iax.conf
1530 * Added Encryption and Trunk status to manager command "iaxpeers"
1534 * The configuration file now holds separate sections for devices and lines.
1535 Please have a look at configs/skinny.conf.sample and change your skinny.conf
1540 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
1541 support for LibOpenR2. http://www.libopenr2.org/
1542 * The UK option waitfordialtone has been added for use with BT analog
1544 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
1545 is used in conjunction with the 'faxdetect' configuration option. When
1546 'faxbuffers' is used and fax tones are detected, the channel will dynamically
1547 switch to the configured faxbuffers policy. For example, to use 6 buffers
1548 and a 'full' buffer policy for a fax transmission, add:
1550 The faxbuffers configuration will be in affect until the call is torn down.
1551 * Added service message support for 4ESS/5ESS switches.
1555 * For DAHDI channels, the CHANNEL() dialplan function now
1556 supports changing the channel's buffer policy (for the current
1557 call only), using this syntax:
1559 exten => s,n,Set(CHANNEL(buffers)=6,full)
1561 This would change the channel to the 'full' buffer policy and
1562 6 (six) buffers. Possible options for this setting are the same
1563 as those in chan_dahdi.conf.
1564 * Added a new dialplan function, CURLOPT, which permits setting various
1565 options that may be useful with the CURL dialplan function, such as
1566 cookies, proxies, connection timeouts, passwords, etc.
1567 * Permit the syntax and synopsis fields of the corresponding dialplan
1568 functions to be individually set from func_odbc.conf.
1569 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
1570 * func_odbc now may specify an insert query to execute, when the write query
1571 affects 0 rows (usually indicating that no such row exists).
1572 * Added a new dialplan function, LISTFILTER, which permits removing elements
1573 from a set list, by name. Uses the same general syntax as the existing CUT
1574 and FIELDQTY dialplan functions, which also manage lists.
1575 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
1576 obtaining realtime data from the dialplan.
1577 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
1578 a subroutine when using the GoSub() and Return() applications.
1579 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
1580 of "core show function AUDIOHOOK_INHERIT" from the CLI
1581 * Added AES_ENCRYPT. For information on its use, please see the output
1582 of "core show function AES_ENCRYPT" from the CLI
1583 * Added AES_DECRYPT. For information on its use, please see the output
1584 of "core show function AES_DECRYPT" from the CLI
1585 * func_odbc now supports database transactions across multiple queries.
1589 * Scheduled meetme conferences may now have their end times extended by
1591 * app_authenticate now gives the ability to select a prompt other than
1593 * app_directory now pays attention to the searchcontexts setting in
1594 voicemail.conf and will look through all contexts, if no context is
1595 specified in the initial argument.
1596 * A new application, Originate, has been introduced, that allows asynchronous
1597 call origination from the dialplan.
1598 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
1599 in addition to the setting in the "general" context.
1600 * Added ConfBridge dialplan application which does conference bridges without
1601 DAHDI. For information on its use, please see the output of
1602 "core show application ConfBridge" from the CLI.
1606 * The Asterisk CLI has a new command, "channel redirect", which is similar in
1607 operation to the AMI Redirect action.
1608 * extensions.conf now allows you to use keyword "same" to define an extension
1609 without actually specifying an extension. It uses exactly the same pattern
1610 as previously used on the last "exten" line. For example:
1611 exten => 123,1,NoOp(something)
1612 same => n,SomethingElse()
1613 * musiconhold.conf classes of type 'files' can now use relative directory paths,
1614 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
1615 * All deprecated CLI commands are removed from the sourcecode. They are now handled
1616 by the new clialiases module. See cli_aliases.conf.sample file.
1617 * Times within timespecs are now accurate down to the minute. This is a change
1618 from historical Asterisk, which only provided timespecs rounded to the nearest
1619 even (read: evenly divisible by 2) minute mark.
1620 * The realtime switch now supports an option flag, 'p', which disables searches for
1622 * In addition to a time range and date range, timespecs now accept a 5th optional
1623 argument, timezone. This allows you to perform time checks on alternate
1624 timezones, especially if those daylight savings time ranges vary from your
1625 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
1627 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
1628 give you the correct output for an asterisk box behind nat. It will give you the
1629 externhost and localnet settings.
1630 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
1631 can connect calls in passthrough mode, as well as record and play back files.
1632 * Successful and unsuccessful call pickup can now be alerted through sounds, by
1633 using pickupsound and pickupfailsound in features.conf.
1634 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
1635 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
1636 instead of the /var/run/asterisk.pid where it used to be. This will make
1637 installs as non-root easier to manage.
1642 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
1643 be written; they will no longer be explicitly written.
1645 Asterisk Manager Interface
1646 --------------------------
1647 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
1648 a non-empty value) in your request. If you do this, any pending AMI events will
1649 *not* be included in the response to your request as they would normally, but
1650 will be left in the event queue for the next request you make to retrieve. For
1651 some applications, this will allow you to guarantee that you will only see
1652 events in responses to 'WaitEvent' actions, and can better know when to expect them.
1653 To know whether the Asterisk server supports this header or not, your client can
1654 inspect the first response back from the server to see if it includes this header:
1656 Pragma: SuppressEvents
1658 If this is included, the server supports event suppression.
1660 * Added 4 new Actions to list skinny device(s) and line(s)
1666 LDAP Schema File Additions
1667 --------------------------
1668 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
1669 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
1671 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
1672 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
1673 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
1674 * Removed redundant IPaddr (there's already IPAddress)
1675 - Gives more configuration Flags for SIP-Users available (tested)
1676 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
1677 without extensibleObject (which really should be the last resort); gives
1678 also additional possibilities for LDAP-filter
1680 ------------------------------------------------------------------------------
1681 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
1682 ------------------------------------------------------------------------------
1684 Device State Handling
1685 ---------------------
1686 * The event infrastructure in Asterisk got another big update to help support
1687 distributed events. It currently supports distributed device state and
1688 distributed Voicemail MWI (Message Waiting Indication). A new module has
1689 been merged, res_ais, which facilitates communicating events between servers.
1690 It uses the SAForum AIS (Service Availability Forum Application Interface
1691 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
1692 a cluster of Asterisk servers, and to share events between them. For more
1693 information on setting this up, refer to the Distributed Device State section
1694 of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1698 * Added a new dialplan function, AST_CONFIG(), which allows you to access
1699 variables from an Asterisk configuration file.
1700 * The JACK_HOOK function now has a c() option to supply a custom client name.
1701 * Added two new dialplan functions from libspeex for audio gain control and
1702 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
1703 rx directions of a channel from the dialplan.
1704 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
1705 based on other parameters. The default is still to search based on the
1706 forwarding station ID. However, there are new options that allow you to search
1707 based on the message desk terminal ID, or the message desk number.
1708 * TIMEOUT() has been modified to be accurate down to the millisecond.
1709 * ENUM*() functions now include the following new options:
1710 - 'u' returns the full URI and does not strip off the URI-scheme.
1711 - 's' triggers ISN specific rewriting
1712 - 'i' looks for branches into an Infrastructure ENUM tree
1713 - 'd' for a direct DNS lookup without any flipping of digits.
1714 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
1715 * CHANNEL() now has options for the maximum, minimum, and standard or normal
1716 deviation of jitter, rtt, and loss for a call using chan_sip.
1718 DAHDI channel driver (chan_dahdi) Changes
1719 ----------------------------------------
1720 * Channels can now be configured using named sections in chan_dahdi.conf, just
1721 like other channel drivers, including the use of templates.
1722 * The default for pridialplan has changed from 'national' to 'unknown'.
1726 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
1727 to something that matches the pattern a hint will be created using the contents
1728 and variables evaluated.
1729 * Dialplan matching has been extended to allow an extension to return to the
1730 PBX core to wait for more digits. This is done by using the new dialplan
1731 application called "Incomplete". This will permit a whole new level of
1732 extension control, by giving the administrator more control over early
1733 matches employing one of the short-circuit pattern match operators. Note
1734 that custom applications can trigger this same behavior by returning the
1735 special value AST_PBX_INCOMPLETE.
1739 * Directory now permits both first and last names to be matched at the same
1740 time. In addition, the number of digits to enter of the name can be set in
1741 the arguments to Directory; previously, you could enter only 3, regardless
1742 of how many names are in your company. For large companies, this should be
1744 * Voicemail now permits a mailbox setting to wrap around from first to last
1745 messages, if the "messagewrap" option is set to a true value.
1746 * Voicemail now permits an external script to be run, for password validation.
1747 The script should output "VALID" or "INVALID" on stdout, depending upon the
1748 wish to validate or invalidate the password given. Arguments are:
1749 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
1751 * Dial has a new option: F(context^extension^pri), which permits a callee to
1752 continue in the dialplan, at the specified label, if the caller hangs up.
1753 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
1754 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
1755 * The Jack application now has a c() option to supply a custom client name.
1756 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
1757 like the pre-existing whisper mode, except that the spy can also talk to the
1758 participant on the bridged channel as well.
1759 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
1760 to be spoken instead of the channel name or number. For more information on the
1761 use of this option, issue the command "core show application ChanSpy" from the
1763 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
1764 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
1765 words, if using the 'd' option, it is not possible to enter a number to append to
1766 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
1767 change to whisper mode, and pressing 6 will change to barge mode.
1768 * ExternalIVR now takes several options that affect the way it performs, as
1769 well as having several new commands. Please see the External IVR page on the Asterisk
1770 wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
1771 * Added ability to communicate over a TCP socket instead of forking a child process for the
1772 ExternalIVR application.
1773 * ChanIsAvail has a new option, 'a', which will return all available channels instead
1774 of just the first one if you give the function more then one channel to check.
1775 * PrivacyManager now takes an option where you can specify a context where the
1776 given number will be matched. This way you have more control over who is allowed
1777 and it stops the people who blindly enter 10 digits.
1778 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
1779 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
1780 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
1781 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
1782 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
1783 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
1784 * The Dial() application no longer copies the language used by the caller to the callee's
1785 channel. If you desire for the caller's channel's language to be used for file playback
1786 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
1787 * SendImage() no longer hangs up the channel on error; instead, it sets the
1788 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
1789 'UNSUPPORTED'. This change makes SendImage() more consistent with other
1791 * Park has a new option, 's', which silences the announcement of the parking space number.
1792 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
1793 invalid input and will be assumed to mean that no timeout is desired.
1797 * Added DNS manager support to registrations for peers referencing peer entries.
1798 DNS manager runs in the background which allows DNS lookups to be run asynchronously
1799 as well as periodically updating the IP address. These properties allow for
1800 better performance as well as recovery in the event of an IP change.
1801 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
1802 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
1803 These changes also provide performance improvements for call setup and tear down.
1804 * Added ability to specify registration expiry time on a per registration basis in
1806 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
1808 * Added t38pt_usertpsource option. See sip.conf.sample for details.
1809 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
1810 * 'sip show peers' and 'sip show users' display their entries sorted in
1811 alphabetical order, as opposed to the order they were in, in the config
1813 * Videosupport now supports an additional option, "always", which always sets
1814 up video RTP ports, even on clients that don't support it. This helps with
1815 callfiles and certain transfers to ensure that if two video phones are
1816 connected, they will always share video feeds.
1820 * Existing DNS manager lookups extended to check for SRV records.
1821 * IAX2 encryption support has been improved to support periodic key rotation
1822 within a call for enhanced security. The option "keyrotate" has been
1823 provided to disable this functionality to preserve backwards compatibility
1824 with older versions of IAX2 that do not support key rotation.
1828 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
1829 data tree based on the given <path>.
1830 * New CLI command "data show providers" that will display all the registered
1832 * New CLI command, "config reload <file.conf>" which reloads any module that
1833 references that particular configuration file. Also added "config list"
1834 which shows which configuration files are in use.
1835 * New CLI commands, "pri show version" and "ss7 show version" that will
1836 display which version of libpri and libss7 are being used, respectively.
1837 A new API call was added so trunk will now have to be compiled against
1838 a versions of libpri and libss7 that have them or it will not know that
1839 these libraries exist.
1840 * The commands "core show globals", "core set global" and "core set chanvar" has
1841 been deprecated in favor of the more semanticly correct "dialplan show globals",
1842 "dialplan set chanvar" and "dialplan set global".
1843 * New CLI command "dialplan show chanvar" to list all variables associated
1844 with a given channel.
1848 * Addresses managed by DNS manager now can check to see if there is a DNS
1849 SRV record for a given domain and will use that hostname/port if present.
1851 AMI - The manager (TCP/TLS/HTTP)
1852 --------------------------------
1853 * The Status command now takes an optional list of variables to display
1854 along with channel status.
1855 * The QueueEntry event now also includes the channel's uniqueid
1859 * res_odbc no longer has a limit of 1023 total possible unshared connections,
1860 as some people were running into this limit. This limit has been increased
1865 * The TRANSFER queue log entry now includes the the caller's original
1866 position in the transferred-from queue.
1867 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
1868 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
1869 as well as an explanation about timeout options in general
1870 * Added a new option - C - for forcing the "answered elsewhere" flag on
1871 cancellation of calls in to members of the queue. This is to avoid the
1872 call to a member of a queue having the call listed as a "missed call".
1876 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
1877 adaptive capabilities. What this means in practical terms is that if your
1878 realtime table lacks critical fields, Asterisk will now emit warnings to
1879 that effect. Also, some of the realtime drivers have the ability (if
1880 configured) to automatically add those columns to the table with the
1881 correct type and length.
1885 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
1886 the 'setvar' option to cause a given audio file to be played upon completion
1887 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
1888 Skinny channels only.
1889 * You can now compile Asterisk against the Hoard Memory Allocator, see the
1890 Hoard page on the Asterisk wiki for more information:
1891 https://wiki.asterisk.org/wiki/x/pQBB
1892 * Config file variables may now be appended to, by using the '+=' append
1893 operator. This is most helpful when working with long SQL queries in
1894 func_odbc.conf, as the queries no longer need to be specified on a single
1896 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
1897 which will add a second to the billsec when the ending
1898 time is set, if the number in the microseconds field of the end time is
1899 greater than the number of microseconds in the answer time. This allows
1900 users to count the 'initiated' seconds in their billing records.
1902 ------------------------------------------------------------------------------
1903 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
1904 ------------------------------------------------------------------------------
1906 AMI - The manager (TCP/TLS/HTTP)
1907 --------------------------------
1908 * Manager has undergone a lot of changes, all of them documented
1909 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
1910 * Manager version has changed to 1.1
1911 * Added a new action 'CoreShowChannels' to list currently defined channels
1912 and some information about them.
1913 * Added a new action 'SIPshowregistry' to list SIP registrations.
1914 * Added TLS support for the manager interface and HTTP server
1915 * Added the URI redirect option for the built-in HTTP server
1916 * The output of CallerID in Manager events is now more consistent.
1917 CallerIDNum is used for number and CallerIDName for name.
1918 * Enable https support for builtin web server.
1919 See configs/http.conf.sample for details.
1920 * Added a new action, GetConfigJSON, which can return the contents of an
1921 Asterisk configuration file in JSON format. This is intended to help
1922 improve the performance of AJAX applications using the manager interface
1924 * SIP and IAX manager events now use "ChannelType" in all cases where we
1925 indicate channel driver. Previously, we used a mixture of "Channel"
1926 and "ChannelDriver" headers.
1927 * Added a "Bridge" action which allows you to bridge any two channels that
1928 are currently active on the system.
1929 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
1930 the voicemail users setup.
1931 * Added 'DBDel' and 'DBDelTree' manager commands.
1932 * cdr_manager now reports events via the "cdr" level, separating it from
1933 the very verbose "call" level.
1934 * Manager users are now stored in memory. If you change the manager account
1935 list (delete or add accounts) you need to reload manager.
1936 * Added Masquerade manager event for when a masquerade happens between
1938 * Added "manager reload" command for the CLI
1939 * Lots of commands that only provided information are now allowed under the
1940 Reporting privilege, instead of only under Call or System.
1941 * The IAX* commands now require either System or Reporting privilege, to
1942 mirror the privileges of the SIP* commands.
1943 * Added ability to retrieve list of categories in a config file.
1944 * Added ability to retrieve the content of a particular category.
1945 * Added ability to empty a context.
1946 * Created new action to create a new file.
1947 * Updated delete action to allow deletion by line number with respect to category.
1948 * Added new action insert to add new variable to category at specified line.
1949 * Updated action newcat to allow new category to be inserted in file above another
1951 * Added new event "JitterBufStats" in the IAX2 channel
1952 * Originate now requires the Originate privilege and, if you want to call out
1953 to a subshell, it requires the System privilege, as well. This was done to
1954 enhance manager security.
1955 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
1956 * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
1957 or manager show command Atxfer from the CLI
1958 * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
1959 details or manager show command IAXregistry from the CLI
1963 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
1964 state in the dialplan, as well as creating custom device states that are
1965 controllable from the dialplan.
1966 * Extend CALLERID() function with "pres" and "ton" parameters to
1967 fetch string representation of calling number presentation indicator
1968 and numeric representation of type of calling number value.
1969 * MailboxExists converted to dialplan function
1970 * A new option to Dial() for telling IP phones not to count the call
1971 as "missed" when dial times out and cancels.
1972 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
1973 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
1974 held for any given channel. Also, locks are automatically freed when a
1976 * Added HINT() dialplan function that allows retrieving hint information.
1977 Hints are mappings between extensions and devices for the sake of
1978 determining the state of an extension. This function can retrieve the list
1979 of devices or the name associated with a hint.
1980 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
1982 * Added SYSINFO() dialplan function which allows retrieval of system information
1983 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
1984 the existence of a dialplan target.
1985 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
1986 upper and lower case, respectively.
1987 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
1988 ID for the call (not the Asterisk call ID or unique ID), provided that the
1989 channel driver supports this. For SIP, you get the SIP call-ID for the
1990 bridged channel which you can store in the CDR with a custom field.
1994 * Added CLI permissions, config file: cli_permissions.conf
1995 default is to allow all commands for every local user/group.
1996 Also this new feature added three new CLI commands:
1997 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
1998 - cli reload permissions
1999 - cli show permissions
2000 * New CLI command "core show hint" (usage: core show hint <exten>)
2001 * New CLI command "core show settings"
2002 * Added 'core show channels count' CLI command.
2003 * Added the ability to set the core debug and verbose values on a per-file basis.
2004 * Added 'queue pause member' and 'queue unpause member' CLI commands
2005 * Ability to set process limits ("ulimit") without restarting Asterisk
2006 * Enhanced "agi debug" to print the channel name as a prefix to the debug
2007 output to make debugging on busy systems much easier.
2008 * New CLI commands "dialplan set extenpatternmatching true/false"
2009 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
2010 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
2011 listed in the startup_commands section of cli.conf will get executed.
2012 * Added a CLI command, "devstate change", which allows you to set custom device
2013 states from the func_devstate module that provides the DEVICE_STATE() function
2014 and handling of the "Custom:" devices.
2015 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
2016 sorted into the different possible callbacks, with the number of entries
2017 currently scheduled for each. Gives you a feel for how busy the sip channel
2019 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
2020 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
2021 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
2025 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
2026 option is enabled, Asterisk will watch for a CNG tone in the incoming audio
2027 for a received call. If it is detected, the channel will jump to the
2028 'fax' extension in the dialplan.
2029 * The default SIP useragent= identifier now includes the Asterisk version
2030 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
2031 If set, and the incoming request carries authentication info,
2032 the username to match in the users list is taken from the Digest header
2033 rather than from the From: field. This feature is considered experimental.
2034 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
2035 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
2036 * The "localmask" setting was removed in version 1.2 and the reminder about it
2037 being removed is now also removed.
2038 * A new option "busylevel" for setting a level of calls where asterisk reports
2039 a device as busy, to separate it from call-limit. This value is also added
2040 to the SIP_PEER dialplan function.
2041 * A new realtime family called "sipregs" is now supported to store SIP registration
2042 data. If this family is defined, "sippeers" will be used for configuration and
2043 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
2044 registration data, as before.
2045 * The SIPPEER function have new options for port address, call and pickup groups
2046 * Added support for T.140 realtime text in SIP/RTP
2047 * The "checkmwi" option has been removed from sip.conf, as it is no longer
2048 required due to the restructuring of how MWI is handled. See the descriptions
2049 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
2050 for more information.
2051 * Added rtpdest option to CHANNEL() dialplan function.
2052 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
2053 * SIP now adds a header to the CANCEL if the call was answered by another phone
2054 in the same dial command, or if the new c option in dial() is used.
2055 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
2056 states it is not needed. For phones, however, that do require it the "registertrying" option
2057 has been added so it can be enabled.
2058 * A new option called "callcounter" (global/peer/user level) enables call counters needed
2059 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
2060 used to enable this functionality).
2061 * New settings for timer T1 and timer B on a global level or per device. This makes it
2062 possible to force timeout faster on non-responsive SIP servers. These settings are
2063 considered advanced, so don't use them unless you have a problem.
2064 * Added a dial string option to be able to set the To: header in an INVITE to any
2066 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
2067 the qualify frequency.
2068 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
2069 were not properly torn down due to network or endpoint failures during an established
2071 * Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB
2072 and configs/sip.conf.sample for more information on how it is used.
2073 * Added a new configuration option "authfailureevents" that enables manager events when
2074 a peer can't authenticate properly.
2075 * Added DNS manager support to registrations for peers not referencing a peer entry.
2079 * Added the trunkmaxsize configuration option to chan_iax2.
2080 * Added the srvlookup option to iax.conf
2081 * Added support for OSP. The token is set and retrieved through the CHANNEL()
2084 XMPP Google Talk/Jingle changes
2085 -------------------------------
2086 * Added the bindaddr option to gtalk.conf.
2090 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
2091 * Proper codec support in chan_skinny.
2092 * Added settings for IP and Ethernet QoS requests
2096 * Added separate settings for media QoS in mgcp.conf
2098 Console Channel Driver changes
2099 ------------------------------
2100 * Added experimental support for video send & receive to chan_oss.
2101 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
2104 Phone channel changes (chan_phone)
2105 ----------------------------------
2106 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
2108 H.323 channel Changes
2109 ---------------------
2110 * H323 remote hold notification support added (by NOTIFY message
2111 and/or H.450 supplementary service)
2113 Local channel changes
2114 ---------------------
2115 * The device state functionality in the Local channel driver has been updated
2116 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
2117 to just UNKNOWN if the extension exists.
2118 * Added jitterbuffer support for chan_local. This allows you to use the
2119 generic jitterbuffer on incoming calls going to Asterisk applications.
2120 For example, this would allow you to use a jitterbuffer for an incoming
2121 SIP call to Voicemail by putting a Local channel in the middle. This
2122 feature is enabled by using the 'j' option in the Dial string to the Local
2123 channel in conjunction with the existing 'n' option for local channels.
2124 * A 'b' option has been added which causes chan_local to return the actual channel
2125 that is behind it when queried. This is useful for transfer scenarios as the
2126 actual channel will be transferred, not the Local channel.
2128 Agent channel changes
2129 ----------------------
2130 * The ackcall and endcall options are now supplemented with options acceptdtmf
2131 and enddtmf. These allow for the DTMF keypress to be configurable. The options
2132 default to their old hard-coded values ('#' and '*' respectively) so this should
2133 not break any existing agent installations.
2135 DAHDI channel driver (chan_dahdi) Changes
2136 ----------------------------------------
2137 * SS7 support (via libss7 library)
2138 * In India, some carriers transmit CID via dtmf. Some code has been added
2139 that will handle some situations. The cidstart=polarity_IN choice has been added for
2140 those carriers that transmit CID via dtmf after a polarity change.
2141 * CID matching information is now shown when doing 'dialplan show'.
2142 * Added dahdi show version CLI command.
2143 * Added setvar support to chan_dahdi.conf channel entries.
2144 * Added two new options: mwimonitor and mwimonitornotify. These options allow
2145 you to enable MWI monitoring on FXO lines. When the MWI state changes,
2146 the script specified in the mwimonitornotify option is executed. An internal
2147 event indicating the new state of the mailbox is also generated, so that
2148 the normal MWI facilities in Asterisk work as usual.
2149 * Added signalling type 'auto', which attempts to use the same signalling type
2150 for a channel as configured in DAHDI. This is primarily designed for analog
2151 ports, but will also work for digital ports that are configured for FXS or FXO
2152 signalling types. This mode is also the default now, so if your chan_dahdi.conf
2153 does not specify signalling for a channel (which is unlikely as the sample
2154 configuration file has always recommended specifying it for every channel) then
2155 the 'auto' mode will be used for that channel if possible.
2156 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
2157 state for a channel; also ensured that the DNDState Manager event is
2158 emitted no matter how the DND state is set or cleared.
2162 * Added a new channel driver, chan_unistim. See the Asterisk wiki at
2163 https://wiki.asterisk.org/wiki/x/vgsiAQ and configs/unistim.conf.sample
2164 for details. This new channel driver allows you to use Nortel i2002,
2165 i2004, and i2050 phones with Asterisk.
2166 * Added a new channel driver, chan_console, which uses portaudio as a cross
2167 platform audio interface. It was written as a channel driver that would
2168 work with Mac CoreAudio, but portaudio supports a number of other audio
2169 interfaces, as well. Note that this channel driver requires v19 or higher
2170 of portaudio; older versions have a different API.
2174 * Added the ability to specify arguments to the Dial application when using
2175 the DUNDi switch in the dialplan.
2176 * Added the ability to set weights for responses dynamically. This can be
2177 done using a global variable or a dialplan function. Using the SHELL()
2178 function would allow you to have an external script set the weight for
2180 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
2181 functions will allow you to initiate a DUNDi query from the dialplan,
2182 find out how many results there are, and access each one.
2183 * Added the ability to specifiy a port for a dundi peer.
2187 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
2188 functions will allow you to initiate an ENUM lookup from the dialplan,
2189 and Asterisk will cache the results. ENUMRESULT can be used to access
2190 the results without doing multiple DNS queries.
2194 * Added the ability to customize which sound files are used for some of the
2195 prompts within the Voicemail application by changing them in voicemail.conf
2196 * Added the ability for the "voicemail show users" CLI command to show users
2197 configured by the dynamic realtime configuration method.
2198 * MWI (Message Waiting Indication) handling has been significantly
2199 restructured internally to Asterisk. It is now totally event based
2200 instead of polling based. The voicemail application will notify other
2201 modules that have subscribed to MWI events when something in the mailbox
2203 This also means that if any other entity outside of Asterisk is changing
2204 the contents of mailboxes, then the voicemail application still needs to
2205 poll for changes. Examples of situations that would require this option
2206 are web interfaces to voicemail or an email client in the case of using
2207 IMAP storage. So, two new options have been added to voicemail.conf
2208 to account for this: "pollmailboxes" and "pollfreq". See the sample
2209 configuration file for details.
2210 * Added "tw" language support
2211 * Added support for storage of greetings using an IMAP server
2212 * Added ability to customize forward, reverse, stop, and pause keys for message playback
2213 * SMDI is now enabled in voicemail using the smdienable option.
2214 * A "lockmode" option has been added to asterisk.conf to configure the file
2215 locking method used for voicemail, and potentially other things in the
2216 future. The default is the old behavior, lockfile. However, there is a
2217 new method, "flock", that uses a different method for situations where the
2218 lockfile will not work, such as on SMB/CIFS mounts.
2219 * Added the ability to backup deleted messages, to ease recovery in the case
2220 that a user accidentally deletes a message, and discovers that they need it.
2221 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
2222 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
2223 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
2224 voicemail boxes. The SMDI interface can also poll for MWI changes when some
2225 outside entity is modifying the state of the mailbox (such as IMAP storage or
2226 a web interface of some kind).
2227 * Added the support for marking messages as "urgent." There are two methods to accomplish
2228 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
2229 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
2230 the message as urgent after he has recorded a voicemail by following the voice instructions.
2231 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
2236 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
2237 used across multiple queues.
2238 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
2239 setqueueentryvar options for each queue, see queues.conf.sample for details.
2240 * Added keepstats option to queues.conf which will keep queue
2241 statistics during a reload.
2242 * setinterfacevar option in queues.conf also now sets a variable
2243 called MEMBERNAME which contains the member's name.
2244 * Added 'Strategy' field to manager event QueueParams which represents
2245 the queue strategy in use.
2246 * Added option to run macro when a queue member is connected to a caller,
2247 see queues.conf.sample for details.
2248 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
2249 does not count paused queue members as unavailable.
2250 * Added min-announce-frequency option to queues.conf which allows you to control the
2251 minimum amount of time between queue announcements for use when the caller's queue
2252 position changes frequently.
2253 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
2255 * Added ability for non-realtime queues to have realtime members
2256 * Added the "linear" strategy to queues.
2257 * Added the "wrandom" strategy to queues.
2258 * Added new channel variable QUEUE_MIN_PENALTY
2259 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
2260 rules in queuerules.conf. See configs/queuerules.conf.sample for details
2261 * Added a new parameter for member definition, called state_interface. This may be
2262 used so that a member may be called via one interface but have a different interface's
2263 device state reported.
2264 * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
2265 "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
2266 "manager show command QueueReset."
2267 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
2268 specified by the periodic-announce option, then one will be chosen randomly when it is time
2269 to play a periodic announcment
2270 * New configuration options: announce-position now takes two more values in addition to "yes" and
2271 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
2272 announce-position-limit. By setting announce-position to "limit" callers will only have their
2273 position announced if their position is less than what is specified by announce-position-limit.
2274 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
2275 will be told that their are more than announce-position-limit callers waiting.
2276 * Two new queue log events have been added. An ADDMEMBER event will be logged
2277 when a realtime queue member is added and a REMOVEMEMBER event will be logged
2278 when a realtime queue member is removed. Since there is no calling channel associated
2279 with these events, the string "REALTIME" is placed where the channel's unique id
2280 is typically placed.
2281 * The configuration method for the "joinempty" and "leavewhenempty" options has
2282 changed to a comma-separated list of methods of determining member availability
2283 instead of vague terms such as "yes," "loose," "no," and "strict." These old four
2284 values are still accepted for backwards-compatibility, though.
2285 * The average talktime is now calculated on queues. This information is reported via the
2286 CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
2287 and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
2292 * The 'o' option to provide an optimization has been removed and its functionality
2293 has been enabled by default.
2294 * When a conference is created, the UNIQUEID of the channel that caused it to be
2295 created is stored. Then, every channel that joins the conference will have the
2296 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
2297 callers that come and go from long standing conferences.
2298 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
2299 except it does operations on a channel by name, instead of number in a conference.
2300 This is a very useful feature in combination with the 'X' option to ChanSpy.
2301 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
2303 * Added new RealTime functionality to provide support for scheduled conferencing.
2304 This includes optional messages to the caller if they attempt to join before
2305 the schedule start time, or to allow the caller to join the conference early.
2306 Also included is optional support for limiting the number of callers per
2307 RealTime conference.
2308 * Added the S() and L() options to the MeetMe application. These are pretty
2309 much identical to the S() and L() options to Dial(). They let you set
2310 timeouts for the conference, as well as have warning sounds played to
2311 let the caller know how much time is left, and when it is running out.
2312 * Added the ability to do "meetme concise" with the "meetme" CLI command.
2313 This extends the concise capabilities of this CLI command to include
2314 listing all conferences, instead of an addition to the other sub commands
2315 for the "meetme" command.
2316 * Added the ability to specify the music on hold class used to play into the
2317 conference when there is only one member and the M option is used.
2318 * Added MEETME_INFO dialplan function which provides a way to query
2319 various properties of a Meetme conference.
2320 * Added new admin features: *81: Roll call, *82: eject all, *83: mute all,
2321 and *84: record in-conf
2323 Other Dialplan Application Changes
2324 ----------------------------------
2325 * Argument support for Gosub application
2326 * From the to-do lists: straighten out the app timeout args:
2327 Wait() app now really does 0.3 seconds- was truncating arg to an int.
2328 WaitExten() same as Wait().
2329 Congestion() - Now takes floating pt. argument.
2330 Busy() - now takes floating pt. argument.
2331 Read() - timeout now can be floating pt.
2332 WaitForRing() now takes floating pt timeout arg.
2333 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
2334 * Added 's' option to Page application.
2335 * Added an optional timeout argument to the Page application.
2336 * Added 'E', 'V', and 'P' commands to ExternalIVR.
2337 * Added 'o' and 'X' options to Chanspy.
2338 * Added a new dialplan application, Bridge, which allows you to bridge the
2339 calling channel to any other active channel on the system.
2340 * Added the ability to specify a music on hold class to play instead of ringing
2341 for the SLATrunk application.
2342 * The Read application no longer exits the dialplan on error. Instead, it sets
2343 READSTATUS to ERROR, which you can catch and handle separately.
2344 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
2345 of asking for verification of each name, one at a time.
2346 * Privacy() no longer uses privacy.conf, as all options are specifyable as
2347 direct options to the app.
2348 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
2350 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
2351 * The ChannelRedirect application no longer exits the dialplan if the given channel
2352 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
2353 or NOCHANNEL if the given channel was not found.
2354 * The silencethreshold setting that was previously configurable in multiple
2355 applications is now settable globally via dsp.conf.
2357 Music On Hold Changes
2358 ---------------------
2359 * A new option, "digit", has been added for music on hold classes in
2360 musiconhold.conf. If this is set for a music on hold class, a caller
2361 listening to music on hold can press this digit to switch to listening
2362 to this music on hold class.
2363 * Support for realtime music on hold has been added.
2364 * In conjunction with the realtime music on hold, a general section has
2365 been added to musiconhold.conf, its sole variable is cachertclasses. If this
2366 is set, then music on hold classes found in realtime will be cached in memory.
2370 * AEL upgraded to use the Gosub with Arguments instead
2371 of Macro application, to hopefully reduce the problems
2372 seen with the artificially low stack ceiling that
2373 Macro bumps into. Macros can only call other Macros
2374 to a depth of 7. Tests run using gosub, show depths
2375 limited only by virtual memory. A small test demonstrated
2376 recursive call depths of 100,000 without problems.
2377 -- in addition to this, all apps that allowed a macro
2378 to be called, as in Dial, queues, etc, are now allowing
2379 a gosub call in similar fashion.
2380 * AEL now generates LOCAL(argname) declarations when it
2381 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
2382 etc. That makes the arguments local in scope. The user
2383 can define their own local variables in macros, now,
2384 by saying "local myvar=someval;" or using Set() in this
2385 fashion: Set(LOCAL(myvar)=someval); ("local" is now
2387 * utils/conf2ael introduced. Will convert an extensions.conf
2388 file into extensions.ael. Very crude and unfinished, but
2389 will be improved as time goes by. Should be useful for a
2390 first pass at conversion.
2391 * aelparse will now read extensions.conf to see if a referenced
2392 macro or context is there before issueing a warning.
2393 * AEL parser sets a local channel variable ~~EXTEN~~, to
2394 preserve the value of ${EXTEN} thru switch statements.
2395 * New operator in $[...] expressions: the ~~ operator serves
2396 as a concatenation operator. AT THE MOMENT, it is really only
2397 necessary and useful in AEL, especially in if() expressions.
2398 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
2399 any enclosing double-quotes, and evaluate to the value of a
2400 concatenated with the value of b. For example if a is set to
2401 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
2402 evaluate to xyzabc .
2405 Call Features (res_features) Changes
2406 ------------------------------------
2407 * Added the parkedcalltransfers option to features.conf
2408 * Added parkedcallparking option to control one touch parking w/ parking
2410 * Added parkedcallhangup option to control disconnect feature w/ parking
2412 * Added parkedcallrecording option to control one-touch record w/ parking
2414 * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
2415 parkedcalltransfers option support for multiple parking lots.
2416 * Added BRIDGE_FEATURES variable to set available features for a channel
2417 * The built-in method for doing attended transfers has been updated to
2418 include some new options that allow you to have the transferee sent
2419 back to the person that did the transfer if the transfer is not successful.
2420 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
2421 in features.conf.sample.
2422 * Added support for configuring named groups of custom call features in
2423 features.conf. This means that features can be written a single time, and
2424 then mapped into groups of features for different key mappings or easier
2426 * Updated the ParkedCall application to allow you to not specify a parking
2427 extension. If you don't specify a parking space to pick up, it will grab
2428 the first one available.
2429 * Added cli command 'features reload' to reload call features from features.conf
2430 * Moved into core asterisk binary.
2431 * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
2432 * Added the ability for custom parking lots to be configured with their own
2433 parking extension with the parkext option.
2435 Language Support Changes
2436 ------------------------
2437 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
2438 * Added support for the Hungarian language for saying numbers, dates, and times.
2442 * Added SPEECH commands for speech recognition. A complete listing can be found
2444 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
2445 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
2446 does not behave as expected; the native command needs to be used, instead.
2447 * Added the ability to perform SRV lookups on fast AGI calls. To use this
2448 feature, simply use hagi: instead of agi: as the protocol portion
2449 of the URI parameter to the AGI function call in your dial plan. Also note
2450 that specifying a port number in the AGI URI will disable SRV lookups,
2451 even if you use the hagi: protocol.
2452 * No longer support MSG_OOB flag on HANGUP.
2456 * Added rotatestrategy option to logger.conf, along with two new options:
2457 "timestamp" which will use the time to name the logger files instead of
2458 sequence number; and "rotate", which rotates the names of the log files,
2459 similar to the way syslog rotates files.
2460 * Added exec_after_rotate option to logger.conf, which allows a system
2461 command to be run after rotation. This is primarily useful with
2462 rotatestrategy=rotate, to allow a limit on the number of log files kept
2463 and to ensure that the oldest log file gets deleted.
2464 * Added realtime support for the queue log
2468 * The cdr_manager module has a [mappings] feature, like cdr_custom,
2469 to add fields to the manager event from the CDR variables.
2470 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
2471 backend database CDR table. Specifically, additional, non-standard
2472 columns are supported, merely by setting the corresponding CDR variable in
2473 your dialplan. In addition, you may alias any column to another name (for
2474 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
2475 simply "alias src => ANI" in the configuration file). Records may be
2476 posted to more than one backend, simply by specifying multiple categories
2477 in the configuration file. And finally, you may filter which CDRs get
2478 posted to each backend, by specifying a filter (which the record must
2479 match) for the particular category. Filters are additive (meaning all
2480 rules must match to post that CDR).
2481 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
2482 module. Specifically, you may add additional columns into the table and
2483 they will be set, if you set the corresponding CDR variable name. Also,
2484 if you omit columns in your database table, they will be silently skipped
2485 (but a record will still be inserted, based on what columns remain). Note
2486 that the other two features from cdr_adaptive_odbc (alias and filter) are
2487 not currently supported.
2488 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
2489 has been disabled using the NoCDR application.
2491 Miscellaneous New Modules
2492 -------------------------
2493 * Added a new CDR module, cdr_sqlite3_custom.
2494 * Added a new realtime configuration module, res_config_sqlite
2495 * Added a new codec translation module, codec_resample, which re-samples
2496 signed linear audio between 8 kHz and 16 kHz to help support wideband
2498 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
2499 based on configuration templates that use Asterisk dialplan function and
2500 variable substitution. It should be possible to create phone profiles and
2501 templates that work for the majority of phones provisioned over http. It
2502 is currently only intended to provision a single user account per phone.
2503 An example profile and set of templates for Polycom phones is provided.
2504 NOTE: Polycom firmware is not included, but should be placed in
2505 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
2506 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
2507 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
2508 provided; there is a JACK() application, and a JACK_HOOK() function. Both
2509 interfaces create an input and output JACK port. The application makes
2510 these ports the endpoint of the call. The audio coming from the channel
2511 goes out the output port and whatever comes back in on the input port is
2512 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
2513 audiohook on the channel. This lets you run the audio coming from a
2514 channel through JACK, and whatever comes back in is what gets forwarded
2515 on as the channel's audio. This is very useful for building custom
2516 vocoders or doing recording or analysis of the channel's audio in another
2518 * Added a new module, res_config_curl, which permits using a HTTP POST url
2519 to retrieve, create, update, and delete realtime information from a remote
2520 web server. Note that this module requires func_curl.so to be loaded for
2521 backend functionality.
2522 * Added a new module, res_config_ldap, which permits the use of an LDAP
2523 server for realtime data access.
2524 * Added support for writing and running your dialplan in lua using the pbx_lua
2525 module. See configs/extensions.lua.sample for examples of how to do this.
2529 * Ability to use libcap to set high ToS bits when non-root
2530 on Linux. If configure is unable to find libcap then you
2531 can use --with-cap to specify the path.
2532 * Added maxfiles option to options section of asterisk.conf which allows you to specify
2533 what Asterisk should set as the maximum number of open files when it loads.
2534 * Added the jittertargetextra configuration option.
2535 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
2536 configuration files for the IP channel drivers. The new option is "cos".
2537 This information is also documented on the Asterisk wiki at
2538 https://wiki.asterisk.org/wiki/x/EYBG
2539 * When originating a call using AMI or pbx_spool that fails the reason for failure
2540 will now be available in the failed extension using the REASON dialplan variable.
2541 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
2542 It allows you to configure a prefix for auto-monitor recordings.
2543 * A new extension pattern matching algorithm, based on a trie, is introduced
2544 here, that could noticeably speed up mid-sized to large dialplans.
2545 It is NOT used by default, as duplicating the behaviour of the old pattern
2546 matcher is still under development. A config file option, in extensions.conf,
2547 in the [general] section, called "extenpatternmatchingnew", is by default
2548 set to false; setting that to true will force the use of the new algorithm.
2549 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
2550 be used to switch the algorithms at run time.
2551 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
2552 specifying which socket to use to connect to the running Asterisk daemon
2554 * Performance enhancements to the sched facility, which is used in
2555 the channel drivers, etc. Added hashtabs and doubly-linked lists
2556 to speed up deletion; start at the beginning or end of list to
2558 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
2559 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
2560 Added regression tests to the tests/ dir, also.
2561 * Added a refcount trace feature to astobj2 for those trying to balance
2562 object creation, deletion; work, play; space and time. See the
2563 notes in astobj2.h. Also, see utils/refcounter as well, as a
2564 quick way to find unbalanced refcounts in what could be a sea
2565 of objects that were balanced.
2566 * Added logging to 'make update' command. See update.log
2567 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
2568 do not come from the remote party.
2569 * Added the 'n' option to the SpeechBackground application to tell it to not
2570 answer the channel if it has not already been answered.
2571 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
2572 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
2574 * iLBC source code no longer included (see UPGRADE.txt for details)
2575 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
2576 deadlock is detected, a backtrace of the stack which led to the lock calls
2577 will be output to the CLI.
2578 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
2579 the "core show locks" CLI command will give lock information output as well
2580 as a backtrace of the stack which led to the lock calls.
2581 * users.conf now sports an optional alternateexts property, which permits
2582 allocation of additional extensions which will reach the specified user.
2583 * A new option for the configure script, --enable-internal-poll, has been added
2584 for use with systems which may have a buggy implementation of the poll system
2585 call. If you notice odd behavior such as the CLI being unresponsive on remote
2586 consoles, you may want to try using this option. This option is enabled by default
2587 on Darwin systems since it is known that the Darwin poll() implementation has
2591 --------------------
2592 * In addition to timing from DAHDI, there is a new timing module called
2593 res_timing_timerfd. In order to use this, you must be running Linux with
2594 a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
2595 script will be able to tell if you have the requirements. From menuselect, select
2596 res_timing_timerfd from the Resource Modules menu.
projects / asterisk/asterisk.git / blob