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 * A new make target, 'full', has been added to the Makefile. This performs
18 the same compilation actions as make all, but will also scan the entirety of
19 each source file for documentation. This option is needed to generate AMI
20 event documentation. Note that your system must have Python in order for
21 this make target to succeed.
25 * The expression parser now recognizes the ABS() absolute value function,
26 which will convert negative floating point values to positive values.
27 * The Asterisk build system will now build and install a shared library
28 (libasteriskssl.so) used to wrap various initialization and shutdown functions
29 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
30 that Asterisk can ensure that these functions do *not* get called by any
31 modules that are loaded into Asterisk, since they should only be called once
32 in any single process. If desired, this feature can be disabled by supplying
33 the "--disable-asteriskssl" option to the configure script.
34 * Threads belonging to a particular call are now linked with callids which get
35 added to any log messages produced by those threads. Log messages can now be
36 easily identified as involved with a certain call by looking at their call id.
37 Call ids may also be attached to log messages for just about any case where
38 it can be determined to be related to a particular call.
39 * The minimum DTMF duration can now be configured in asterisk.conf
40 as "mindtmfduration". The default value is (as before) set to 80 ms.
41 (previously it was only available in source code)
42 * Each logging destination and console now have an independent notion of the
43 current verbosity level. Logger.conf now allows an optional argument to
44 the 'verbose' specifier, indicating the level of verbosity sent to that
45 particular logging destination. Additionally, remote consoles now each
46 have their own verbosity level. The command 'core set verbose' will now set
47 a separate level for each remote console without affecting any other
52 * mixmonitor list <channel> command will now show MixMonitor ID, and the filenames
53 of all running mixmonitors on a channel.
54 * The debuglevel of "pri set debug" is now a bitmask ranging from 0 to 15 if
55 numeric instead of 0, 1, or 2.
59 * Added menu action admin_toggle_mute_participants. This will mute / unmute
60 all non-admin participants on a conference. The confbridge configuration file
61 also allows for the default sounds played to all conference users when this
62 occurs to be overriden using sound_participants_unmuted and sound_participants_muted.
63 * Added menu action participant_count. This will playback the number of current
64 participants in a conference.
65 * Added announcement configuration option to user profile. If set the sound file will
66 be played to the user, and only the user, upon joining the conference bridge.
70 * Addition of the VM_INFO function - see Dialplan function changes
71 * The imapserver, imapport, and imapflags configuration options can now be
72 overriden on a user by user basis.
76 * Asterisk will no longer substitute CID number for CID name into display
77 name field if CID number exists without a CID name. This change improves
78 compatibility with certain device features such as Avaya IP500's directory
80 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
81 created using that setting to not be removed during SIP reload.
82 * Add support to realtime for the 'callbackextension' option
83 * When multiple peers exist with the same address, but differing
84 callbackextension options, incoming requests that are matched by address
85 will be matched to the peer with the matching callbackextension if it is
87 * NAT settings are now a combinable list of options. The equivalent of the
88 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
89 * Two new NAT options, auto_force_rport and auto_comedia, have been added
90 which set the force_rport and comedia options automatically if Asterisk
91 detects that an incoming SIP request crossed a NAT after being sent by
93 * Adds an option send_diversion which can be disabled to prevent
94 diversion headers from automatically being added to invites.
95 * Add support for lightweight NAT keepalive. If enabled a blank packet will
96 be sent to the remote host at a given interval to keep the NAT mapping open.
97 This can be enabled using the keepalive configuration option.
101 * Added a manager event "LocalBridge" for local channel call bridges between
102 the two pseudo-channels created.
106 * Added dialtone_detect option for analog ports to disconnect incoming
107 calls when dialtone is detected.
111 * Added ability to use multiple lines on phone, so for one device in
112 configuration multiple lines can be defined, it allows to have multiple calls
113 on one phone, callwaiting and switching between calls.
114 * Added option 'sharpdial' allowing end dialing by pressing # key
115 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
116 * Added global 'debug' option, that enables debug in channel driver
117 * Added ability for translation on-screen menu to multiple languages. Tested on
118 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
119 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
121 * Reworked dialing number input: added dialing by timeout, immediate dial on
122 on dialplan compare, phone number length now not limited by screen size
123 * Added ability for pickup a call using fetures.conf defined value and
128 * Codec lists may now be modified by the '!' character, to allow succinct
129 specification of a list of codecs allowed and disallowed, without the
130 requirement to use two different keywords. For example, to specify all
131 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
133 Music On Hold Changes
134 ---------------------
135 * Added 'announcement' option which will play at the start of MOH and between
136 songs in modes of MOH that can detect transitions between songs (eg.
141 * Added queue options autopausebusy and autopauseunavail for automatically
142 pausing a queue member when their device reports busy or congestion.
143 * The 'ignorebusy' option for queue members has been deprecated in favor of
144 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
145 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
146 per interface basis. Individual ringinuse values can now be set in
147 queues.conf via an argument to member definitions. Lastly, the queue
148 'ringinuse' setting now only determines defaults for the per member
149 'ringinuse' setting and does not override per member settings like it does
154 * When voicemail plays a message's envelope with saycid set to yes, when reaching
155 the caller id field it will play a recording of a file with the same base name
156 as the sender's callerid if there is a similarly named file in
157 <astspooldir>/recordings/callerids/
161 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
162 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
163 changed arguments to SayUnixTime so that every option is truly optional even
164 when using multiple options (so that j option could be used without having to
165 manually specify timezone and format) There are other beneftis eg. format can
166 now be used without specifying time zone as well.
167 * Added 'F()' option to Queue and Bridge. Similar to the dial option, these can
168 be supplied with arguments indicating where the callee should go after the caller
169 is hung up, or without options specified, the priority after the Queue/Bridge
171 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
172 channels respectively before the callee channels are called.
176 * New per parking lot options: comebackcontext and comebackdialtime. See
177 configs/features.conf.sample for more details.
179 * Channel variable PARKER is now set when comebacktoorigin is disabled in
182 * MixMonitor hooks now have IDs associated with them which can be used to assign
183 a target to StopMixMonitor. Use of MixMonitor's i(variable) option will allow
184 storage of the MixMontior ID in a channel variable. StopMixmonitor now accepts
185 that ID as an argument.
187 CDR postgresql driver changes
188 -----------------------------
189 * Added command "cdr show pgsql status" to check connection status
191 AMI (Asterisk Manager Interface) changes
192 ----------------------------------------
193 * Originate now generates an error response if the extension given
194 is not found in the dialplan
196 * MixMonitor will now show IDs associated with the mixmonitor upon creating them
197 if the i(variable) option is used. StopMixMonitor will accept MixMonitorID as
198 on option to close specific MixMonitors.
200 * The SIPshowpeer manager action response field "SIP-Forcerport" has been updated
201 to include information about peers configured with nat=auto_force_rport by
202 returning "A" if auto_force_rport is set and nat is detected, and "a" if it is
203 set and nat is not detected. "Y" and "N" are still returned if auto_force_rport
206 * Hangup now can take a regular expression as the Channel option. If you want
207 to hangup multiple channels, use /regex/ as the Channel option. Existing
208 behavior to hanging up a single channel is unchanged, but if you pass a regex,
209 the manager will send you a list of channels back that were hung up.
211 * Support for IPv6 addresses has been added.
213 * AMI Events can now be documented in the Asterisk source. Two new CLI
214 commands have been added to display information about AMI events at run time:
215 manager show events, which shows a list of all known and documented AMI
216 events, and manager show event [event name], which shows detail information
217 about a specific AMI event. Note that AMI event documentation is only
218 generated when Asterisk is compiled using 'make full'.
222 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
223 control of faxdetect.
227 * Allow the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to be
228 used within the dynamic weight attribute when specifying a mapping.
232 * Addition of the VM_INFO function that can be used to retrieve voicemail
233 user information, such as the email address and full name.
234 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
236 * The REDIRECTING function now supports the redirecting original party id
238 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
239 lets you set some of the configuration options from the [general] section
240 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
241 the key sequence used to activate built-in features, such as blindxfer,
242 and automon. See the built-in documentation for details.
246 * A new option, 'I' has been added to app_followme.
247 By setting this option, Asterisk will not update the caller with
248 connected line changes when they occur. This is similar to app_dial
250 * The 'N' option is now ignored if the call is already answered.
251 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
252 and caller channels respectively before the callee channels are called.
256 * A new option, 'probation' has been added to rtp.conf
257 RTP in strictrtp mode can now require more than 1 packet to exit learning
258 mode with a new source (and by default requires 4). The probation option
259 allows the user to change the required number of packets in sequence to any
260 desired value. Use a value of 1 to essentially restore the old behavior.
261 Also, with strictrtp on, Asterisk will now drop all packets until learning
262 mode has successfully exited. These changes are based on how pjmedia handles
263 media sources and source changes.
267 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
268 instead of simply the uri. This is the format that MessageSend() can use
269 in the from parameter for outgoing SIP messages.
273 * A new module, res_corosync, has been introduced. This module uses the
274 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
275 of Asterisk servers to both Message Waiting Indication (MWI) and/or
276 Device State (presence) information. This module is very similar to, and
277 is a replacement for the res_ais module that was in previous releases of
282 * A new channel variable, AGIEXITONHANGUP, has been added which allows
283 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
284 AGI application would exit immediately after a channel hangup is detected.
285 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
286 are resolved and each address is attempted in turn until one succeeds or
289 ------------------------------------------------------------------------------
290 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
291 ------------------------------------------------------------------------------
295 * Asterisk now has protocol independent support for processing text messages
296 outside of a call. Messages are routed through the Asterisk dialplan.
297 SIP MESSAGE and XMPP are currently supported. There are options in
298 jabber.conf and sip.conf to allow enabling these features.
299 -> jabber.conf: see the "sendtodialplan" and "context" options.
300 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
301 and "outofcall_message_context" options.
302 The MESSAGE() dialplan function and MessageSend() application have been
303 added to go along with this functionality. More detailed usage information
304 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
305 * If real-time text support (T.140) is negotiated, it will be preferred for
306 sending text via the SendText application. For example, via SIP, messages
307 that were once sent via the SIP MESSAGE request would be sent via RTP if
308 T.140 text is negotiated for a call.
312 * parkedmusicclass can now be set for non-default parking lots.
314 Asterisk Manager Interface
315 --------------------------
316 * PeerStatus now includes Address and Port.
317 * Added Hold events for when the remote party puts the call on and off hold
318 for chan_dahdi ISDN channels.
319 * Added new action MeetmeListRooms to list active conferences (shows same
320 data as "meetme list" at the CLI).
321 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
322 Description field that is set by 'description' in the channel configuration
324 * Added Uniqueid header to UserEvent.
325 * Added new action FilterAdd to control event filters for the current session.
326 This requires the system permission and uses the same filter syntax as
327 filters that can be defined in manager.conf
328 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
329 versions had some instances of the event converted, but others were left
330 as-is. All Unlink events should now be converted to Bridge events. The AMI
331 protocol version number was incremented to 1.2 as a result of this change.
334 --------------------------
335 * The HTTP Server can bind to IPv6 addresses.
338 --------------------------
339 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
340 with busydetect. usage example: busypattern=200,200,200,600
343 --------------------------
344 * New 'gtalk show settings' command showing the current settings loaded from
346 * The 'logger reload' command now supports an optional argument, specifying an
347 alternate configuration file to use.
348 * 'dialplan add extension' command will now automatically create a context if
349 the specified context does not exist with a message indicated it did so.
350 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
351 Description field which can be populated with 'description' in the channel
352 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
355 --------------------------
356 * The filter option in cdr_adaptive_odbc now supports negating the argument,
357 thus allowing records which do NOT match the specified filter.
358 * Added ability to log CONGESTION calls to CDR
361 --------------------------
362 * Ability to define custom SILK formats in codecs.conf.
363 * Addition of speex32 audio format with translation.
364 * CELT codec pass-through support and ability to define
365 custom CELT formats in codecs.conf.
366 * Ability to read raw signed linear files with sample rates
367 ranging from 8khz - 192khz. The new file extensions introduced
368 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
369 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
370 Skinny, H.323, etc) can still only support the following codecs:
371 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
372 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
373 Video: h261, h263, h263p, h264, mpeg4
378 --------------------------
379 * New highly optimized and customizable ConfBridge application capable of
380 mixing audio at sample rates ranging from 8khz-96khz.
381 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
382 and bridge profiles on a channel.
383 * CONFBRIDGE_INFO dialplan function capable of retrieving information
384 about a conference such as locked status and number of parties, admins,
386 * Addition of video_mode option in confbridge.conf for adding video support
387 into a bridge profile.
388 * Addition of the follow_talker video_mode in confbridge.conf. This video
389 mode dynamically switches the video feed to always display the loudest talker
390 supplying video in the conference.
394 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
395 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
396 variables from asterisk.conf.
400 * Addition of the JITTERBUFFER dialplan function. This function allows
401 for jitterbuffering to occur on the read side of a channel. By using
402 this function conference applications such as ConfBridge and MeetMe can
403 have the rx streams jitterbuffered before conference mixing occurs.
404 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
406 * Added STRREPLACE function. This function let's the user search a variable
407 for a given string to replace with another string as many times as the
408 user specifies or just throughout the whole string.
409 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
410 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
411 * Added extensions to chan_ooh323 in function CHANNEL()
413 libpri channel driver (chan_dahdi) DAHDI changes
414 --------------------------
415 * Added moh_signaling option to specify what to do when the channel's bridged
416 peer puts the ISDN channel on hold.
417 * Added display_send and display_receive options to control how the display ie
418 is handled. To send display text from the dialplan use the SendText()
419 application when the option is enabled.
420 * Added mcid_send option to allow sending a MCID request on a span.
423 --------------------------
424 * Added setvar option to calendar.conf to allow setting channel variables on
425 notification channels.
426 * Added "calendar show types" CLI command to list registered calendar
430 --------------------------
431 * Added two new options, r and t with file name arguments to record
432 single direction (unmixed) audio recording separate from the bidirectional
433 (mixed) recording. The mixed file name argument is optional now as long
434 as at least one recording option is used.
437 --------------------------
438 * Added a new option, l, which will disable local call optimization for
439 channels involved with the FollowMe thread. Use this option to improve
440 compatability for a FollowMe call with certain dialplan apps, options, and
444 --------------------------
445 * Added option "k" that will automatically close the conference when there's
446 only one person left when a user exits the conference.
449 --------------------------
450 * cel_pgsql now supports the 'extra' column for data added using the
451 CELGenUserEvent() application.
454 --------------------------
455 * Support for defining hints has been added to pbx_lua. See the 'hints' table
456 in the sample extensions.lua file for syntax details.
457 * Applications that perform jumps in the dialplan such as Goto will now
458 execute properly. When pbx_lua detects that the context, extension, or
459 priority we are executing on has changed it will immediately return control
460 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
461 the priority after the currently executing priority.
462 * An autoservice is now started by default for pbx_lua channels. It can be
463 stopped and restarted using the autoservice_stop() and autoservice_start()
467 --------------------------
468 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
469 into a FAXStatus event with an 'Operation' header that will be either
470 'send', 'receive', and 'gateway'.
471 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
472 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
473 feature will handle converting a fax call between an audio T.30 fax terminal
474 and an IFP T.38 fax terminal.
478 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
479 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
480 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
484 * Added general option negative_penalty_invalid default off. when set
485 members are seen as invalid/logged out when there penalty is negative.
486 for realtime members when set remove from queue will set penalty to -1.
487 * Added queue option autopausedelay when autopause is enabled it will be
488 delayed for this number of seconds since last successful call if there
489 was no prior call the agent will be autopaused immediately.
490 * Added member option ignorebusy this when set and ringinuse is not
491 will allow per member control of multiple calls as ringinuse does for
493 * Added global option check_state_unknown to enforce checking of device state
494 when the device state is unknown app_queue will see unknown as available.
498 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
500 * Added 'k' option to MeetMe to automatically kill the conference when there's only
501 one participant left (much like a normal call bridge)
502 * Added extra argument to Originate to set timeout.
506 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
507 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
508 utility in the UTILS section of menuselect. If an existing astdb is found and no
509 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
510 convert an existing astdb to the SQLite3 version automatically at runtime.
514 * Modules marked as deprecated are no longer marked as building by default. Enabling
515 these modules is still available via menuselect.
519 * authdebug is now disabled by default. To enable this functionaility again
520 set authdebug = yes in iax.conf.
524 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
525 releases it was disabled.
529 * The PBX core previously made a call with a non-existing extension test for
530 extension s@default and jump there if the extension existed.
531 This was a bad default behaviour and violated the principle of least surprise.
532 It has therefore been changed in this release. It may affect some
533 applications and configurations that rely on this behaviour. Most channel
534 drivers have avoided this for many releases by testing whether the extension
535 called exists before starting the PBX and generating a local error.
536 This behaviour still exists and works as before.
538 Extension "s" is used when no extension is given in a channel driver,
539 like immediate answer in DAHDI or calling to a domain with no user part
542 ------------------------------------------------------------------------------
543 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
544 ------------------------------------------------------------------------------
548 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
549 now defaults to force_rport. It is very important that phones requiring nat=no be
550 specifically set as such instead of relying on the default setting. If at all
551 possible, all devices should have nat settings configured in the general section as
552 opposed to configuring nat per-device.
553 * Added preferred_codec_only option in sip.conf. This feature limits the joint
554 codecs sent in response to an INVITE to the single most preferred codec.
555 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
556 to be used for the outgoing call. It must be one of the codecs configured
558 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
559 to be used for holding a private key. If tlsprivatekey is not specified,
560 tlscertfile is searched for both public and private key.
561 * Added tlsclientmethod option to sip.conf. This allows the protocol for
562 outbound client connections to be specified.
563 * The sendrpid parameter has been expanded to include the options
564 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
565 header to be sent (equivalent to setting sendrpid=yes) and setting
566 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
567 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
568 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
569 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
570 will accept the SDP even if the SDP version number is not properly incremented,
571 but will generate a warning in the log indicating that the SIP peer that sent
572 the SDP should have the 'ignoresdpversion' option set.
573 * The 'nat' option has now been been changed to have yes, no, force_rport, and
574 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
575 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
576 remote side requests it and disables symmetric RTP support. Setting it to
577 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
578 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
579 and enables symmetric RTP support.
580 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
581 response. This permits the master channel to know how each channel dialled
582 in a multi-channel setup resolved in an individual way. This carries a
583 performance penalty and can be disabled in sip.conf using the
584 'storesipcause' option.
585 * Added 'externtcpport' and 'externtlsport' options to allow custom port
586 configuration for the externip and externhost options when tcp or tls is used.
587 * Added support for message body (stored in content variable) to SIP NOTIFY message
588 accessible via AMI and CLI.
589 * Added 'media_address' configuration option which can be used to explicitly specify
590 the IP address to use in the SDP for media (audio, video, and text) streams.
591 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
592 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
594 * Added 'use_q850_reason' configuration option for generating and parsing
595 if available Reason: Q.850;cause=<cause code> header. It is implemented
596 in some gateways for better passing PRI/SS7 cause codes via SIP.
597 * When dialing SIP peers, a new component may be added to the end of the dialstring
598 to indicate that a specific remote IP address or host should be used when dialing
599 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
600 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
601 ability to selectively force bridged channels to also be encrypted is also
602 implemented. Branching in the dialplan can be done based on whether or not
603 a channel has secure media and/or signaling.
604 * Added directmediapermit/directmediadeny to limit which peers can send direct media
606 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
607 Charge messages to snom phones.
608 * Added support for G.719 media streams.
609 * Added support for 16khz signed linear media streams.
610 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
611 RTP has been outfitted with the same abilities.
612 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
613 available in device configurations as well as in the dial plan.
614 * Addition of the 'subscribe_network_change' option for turning on and off
615 res_stun_monitor module support in chan_sip.
616 * Addition of the 'auth_options_requests' option for turning on and off
617 authentication for OPTIONS requests in chan_sip.
621 * Add #tryinclude statement for config files. This provides the same
622 functionality as the #include statement however an asterisk module will
623 still load if the filename does not exist. Using the #include statement
624 Asterisk will not allow the module to load.
628 * Added rtsavesysname option into iax.conf to allow the systname to be saved
630 * Added the ability for chan_iax2 to inform the dialplan whether or not
631 encryption is being used. This interoperates with the SIP SRTP implementation
632 so that a secure SIP call can be bridged to a secure IAX call when the
633 dialplan requires bridged channels to be "secure".
634 * Addition of the 'subscribe_network_change' option for turning on and off
635 res_stun_monitor module support in chan_iax.
640 * Added ability to preset channel variables on indicated lines with the setvar
641 configuration option. Also, clearvars=all resets the list of variables back
643 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
644 See configs/res_pktccops.conf for more information.
646 XMPP Google Talk/Jingle changes
647 -------------------------------
648 * Added the externip option to gtalk.conf.
649 * Added the stunaddr option to gtalk.conf which allows for the automatic
650 retrieval of the external ip from a stun server.
654 * Added 'p' option to PickupChan() to allow for picking up channel by the first
655 match to a partial channel name.
656 * Added .m3u support for Mp3Player application.
657 * Added progress option to the app_dial D() option. When progress DTMF is
658 present, those values are sent immediately upon receiving a PROGRESS message
659 regardless if the call has been answered or not.
660 * Added functionality to the app_dial F() option to continue with execution
661 at the current location when no parameters are provided.
662 * Added the 'a' option to app_dial to answer the calling channel before any
663 announcements or macros are executed.
664 * Modified app_dial to set answertime when the called channel answers even if
665 the called channel hangs up during playback of an announcement.
666 * Modified app_dial 'r' option to support an additional parameter to play an
667 indication tone from indications.conf
668 * Added c() option to app_chanspy. This option allows custom DTMF to be set
669 to cycle through the next available channel. By default this is still '*'.
670 * Added x() option to app_chanspy. This option allows DTMF to be set to
671 exit the application.
672 * The Voicemail application has been improved to automatically ignore messages
673 that only contain silence.
674 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
675 associated mailbox(es) to be greetings-only.
676 * The ChanSpy application now has the 'S' option, which makes the application
677 automatically exit once it hits a point where no more channels are available
679 * The ChanSpy application also now has the 'E' option, which spies on a single
680 channel and exits when that channel hangs up.
681 * The MeetMe application now turns on the DENOISE() function by default, for
682 each participant. In our tests, this has significantly decreased background
683 noise (especially noisy data centers).
684 * Voicemail now permits storage of secrets in a separate file, located in the
685 spool directory of each individual user. The control for this is located in
686 the "passwordlocation" option in voicemail.conf. Please see the sample
687 configuration for more information.
688 * The ChanIsAvail application now exposes the returned cause code using a separate
689 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
690 * Added 'd' option to app_followme. This option disables the "Please hold"
692 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
693 received will terminate recording.
694 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
695 Previously the folder could only be set per context, but has now been extended
696 using the imapfolder option.
697 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
698 * Voicemail now allows the pager date format to be specified separately from the
700 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
701 to allow joining, leaving, and sending text to group chats.
702 * MeetMe has a new option 'G' to play an announcement before joining a conference.
703 * Page has a new option 'A(x)' which will playback an announcement simultaneously
704 to all paged phones (and optionally excluding the caller's one using the new
705 option 'n') before the call is bridged.
706 * The 'f' option to Dial has been augmented to take an optional argument. If no
707 argument is provided, the 'f' option works as it always has. If an argument is
708 provided, then the connected party information of all outgoing channels created
709 during the Dial will be set to the argument passed to the 'f' option.
710 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
712 * The OSP lookup application adds in/outbound network ID, optional security,
713 number portability, QoS reporting, destination IP port, custom info and service
715 * Added new application VMSayName that will play the recorded name of the voicemail
716 user if it exists, otherwise will play the mailbox number.
717 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
718 retrieve state for a particular bridge, where <name> is the conference name
719 * app_directory now allows exiting at any time using the operator or pound key.
720 * Voicemail now supports setting a locale per-mailbox.
721 * Two new applications are provided for declining counting phrases in multiple
722 languages. See the application notes for SayCountedNoun and SayCountedAdj for
724 * Voicemail now runs the externnotify script when pollmailboxes is activated and
726 * Voicemail now includes rdnis within msgXXXX.txt file.
727 * ExternalIVR now supports IPv6 addresses.
728 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
729 at https://wiki.asterisk.org/wiki/x/oQBB
730 * ParkedCall and Park can now specify the parking lot to use.
734 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
735 over SRV records associated with a specific service. From the CLI, type
736 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
737 details on how these may be used.
738 * PITCH_SHIFT dialplan function added. This function can be used to modify the
739 pitch of a channel's tx and rx audio streams.
740 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
741 setting various connected line and redirecting party information.
742 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
743 support ISDN subaddressing.
744 * The CHANNEL() function now supports the "name" and "checkhangup" options.
745 * For DAHDI channels, the CHANNEL() dialplan function now allows
746 the dialplan to request changes in the configuration of the active
747 echo canceller on the channel (if any), for the current call only.
750 exten => s,n,Set(CHANNEL(echocan_mode)=off)
752 The possible values are:
754 on - normal mode (the echo canceller is actually reinitialized)
756 fax - FAX/data mode (NLP disabled if possible, otherwise completely
758 voice - voice mode (returns from FAX mode, reverting the changes that
759 were made when FAX mode was requested)
760 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
761 and setting variables on the channel which created the current channel.
762 Administrators should take care to avoid naming conflicts, when multiple
763 channels are dialled at once, especially when used with the Local channel
764 construct (which all could set variables on the master channel). Usage
765 of the HASH() dialplan function, with the key set to the name of the slave
766 channel, is one approach that will avoid conflicts.
767 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
769 * func_odbc now allows multiple row results to be retrieved without using
770 mode=multirow. If rowlimit is set, then additional rows may be retrieved
771 from the same query by using the name of the function which retrieved the
772 first row as an argument to ODBC_FETCH().
773 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
774 dialplan. This function returns the content of the received message.
775 * Added REPLACE, which searches a given variable name for a set of characters,
776 then either replaces them with a single character or deletes them.
777 * Added PASSTHRU, which literally passes the same argument back as its return
778 value. The intent is to be able to use a literal string argument to
779 functions that currently require a variable name as an argument.
780 * HASH-associated variables now can be inherited across channel creation, by
781 prefixing the name of the hash at assignment with the appropriate number of
782 underscores, just like variables.
783 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
784 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
785 whether or not channels that are bridged to the current channel will be
786 required to have secure signaling and/or media.
787 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
788 the current channel has secure signaling and/or media.
789 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
790 "no_media_path" option.
791 Returns "0" if there is a B channel associated with the call.
792 Returns "1" if no B channel is associated with the call. The call is either
793 on hold or is a call waiting call.
794 * Added option to dialplan function CDR(), the 'f' option
795 allows for high resolution times for billsec and duration fields.
796 * FILE() now supports line-mode and writing.
797 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
798 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
802 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
803 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
804 and is set when a dynamic feature is triggered.
805 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
806 to dynamically create a new parking lot matching the value this varible is
808 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
809 features.conf that should be the base for dynamic parkinglots.
810 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
811 parkinglot should have.
812 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
813 parkinglot should have.
814 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
819 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
821 * Added 'R' option to app_queue. This option stops moh and indicates ringing
822 to the caller when an Agent's phone is ringing. This can be used to indicate
823 to the caller that their call is about to be picked up, which is nice when
824 one has been on hold for an extened period of time.
825 * A new config option, penaltymemberslimit, has been added to queues.conf.
826 When set this option will disregard penalty settings when a queue has too
828 * A new option, 'I' has been added to both app_queue and app_dial.
829 By setting this option, Asterisk will not update the caller with
830 connected line changes or redirecting party changes when they occur.
831 * A 'relative-periodic-announce' option has been added to queues.conf. When
832 enabled, this option will cause periodic announce times to be calculated
833 from the end of announcements rather than from the beginning.
834 * The autopause option in queues.conf can be passed a new value, "all." The
835 result is that if a member becomes auto-paused, he will be paused in all
836 queues for which he is a member, not just the queue that failed to reach
838 * Added dialplan function QUEUE_EXISTS to check if a queue exists
839 * The queue logger now allows events to optionally propagate to a file,
840 even when realtime logging is turned on. Additionally, realtime logging
841 supports sending the event arguments to 5 individual fields, although it
842 will fallback to the previous data definition, if the new table layout is
845 mISDN channel driver (chan_misdn) changes
846 ----------------------------------------
847 * Added display_connected parameter to misdn.conf to put a display string
848 in the CONNECT message containing the connected name and/or number if
849 the presentation setting permits it.
850 * Added display_setup parameter to misdn.conf to put a display string
851 in the SETUP message containing the caller name and/or number if the
852 presentation setting permits it.
853 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
854 indicate the dialplan settings are to be obtained from the asterisk
856 * Made misdn.conf parameter callerid accept the "name" <number> format
857 used by the rest of the system.
858 * Made use the nationalprefix and internationalprefix misdn.conf
859 parameters to prefix any received number from the ISDN link if that
860 number has the corresponding Type-Of-Number. NOTE: This includes
861 comparing the incoming call's dialed number against the MSN list.
862 * Added the following new parameters: unknownprefix, netspecificprefix,
863 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
864 received number from the ISDN link if that number has the corresponding
866 * Added new dialplan application misdn_command which permits controlling
867 the CCBS/CCNR functionality.
868 * Added new dialplan function mISDN_CC which permits retrieval of various
869 values from an active call completion record.
870 * For PTP, you should manually send the COLR of the redirected-to party
871 for an incomming redirected call if the incoming call could experience
872 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
873 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
874 if the REDIRECTING(from-num) is not empty.
875 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
876 option on all of the REDIRECTING statements before dialing the
877 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
878 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
879 redirecting-to presentation (COLR) when it becomes available.
880 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
883 thirdparty mISDN enhancements
884 -----------------------------
885 mISDN has been modified by Digium, Inc. to greatly expand facility message
887 * Enhanced COLP support for call diversion and transfer.
890 The latest modified mISDN v1.1.x based version is available at:
891 http://svn.digium.com/svn/thirdparty/mISDN/trunk
892 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
894 Tagged versions of the modified mISDN code are available under:
895 http://svn.digium.com/svn/thirdparty/mISDN/tags
896 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
898 libpri channel driver (chan_dahdi) DAHDI changes
899 -------------------------------------------
900 * The channel variable PRIREDIRECTREASON is now just a status variable
901 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
902 to read and alter the reason.
903 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
904 redirected-to party for an incomming redirected call if the incoming call
905 could experience further redirects. Just set the
906 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
907 to the COLR. A call has been redirected if the REDIRECTING(count) is not
909 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
910 use the inhibit(i) option on all of the REDIRECTING statements before
911 dialing the redirected-to party. You still have to set the
912 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
913 will update the redirecting-to presentation (COLR) when it becomes available.
914 * Added the ability to ignore calls that are not in a Multiple Subscriber
915 Number (MSN) list for PTMP CPE interfaces.
916 * Added dynamic range compression support for dahdi channels. It is
917 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
918 * Added support for ISDN calling and called subaddress with partial support
919 for connected line subaddress.
920 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
921 * Added handling of received HOLD/RETRIEVE messages and the optional ability
922 to transfer a held call on disconnect similar to an analog phone.
923 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
924 Will reroute/deflect an outgoing call when receive the message.
925 Can use the DAHDISendCallreroutingFacility to send the message for the
927 * Added standard location to add options to chan_dahdi dialing:
928 Dial(DAHDI/g1[/extension[/options]])
931 R Reverse charging indication
932 * Added Reverse Charging Indication (Collect calls) send/receive option.
933 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
934 Dial(DAHDI/g1/extension/R)
935 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
936 (requires latest LibPRI)
937 * Added ability to send/receive keypad digits in the SETUP message.
938 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
939 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
940 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
941 (requires latest LibPRI)
942 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
943 to eliminate tromboned calls. A tromboned call goes out an interface and comes
944 back into the same interface. Tromboned calls happen because of call routing,
945 call deflection, call forwarding, and call transfer.
946 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
947 * Added the ability to support call waiting calls. (The SETUP has no B channel
949 * Added Malicious Call ID (MCID) event to the AMI call event class.
950 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
952 Asterisk Manager Interface
953 --------------------------
954 * The Hangup action now accepts a Cause header which may be used to
955 set the channel's hangup cause.
956 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
957 to specify a separate .pem file to hold a private key. By default sslcert
958 is used to hold both the public and private key.
959 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
960 for options containing the 'tls' prefix. For example, 'sslenable' is now
961 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
962 across all .conf files. All affected sample.conf files have been modified to
963 reflect this change. Previous options such as 'sslenable' still work,
964 but options with the 'tls' prefix are preferred.
965 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
966 in a channel. (res_mutestream.so)
967 * The configuration file manager.conf now supports a channelvars option, which
968 specifies a list of channel variables to include in each channel-oriented
970 * The redirect command now has new parameters ExtraContext, ExtraExtension,
971 and ExtraPriority to allow redirecting the second channel to a different
972 location than the first.
973 * Added new event "JabberStatus" in the Jabber module to monitor buddies
975 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
976 in a MixMonitor recording.
977 * The 'iax2 show peers' output is now similar to the expected output of
979 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
981 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
982 AOC-E messages on a channel.
983 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
984 conform more closely to similar events.
985 * Added a new eventfilter option per user to allow whitelisting and blacklisting
987 * Added optional parkinglot variable for park command.
988 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
989 if CallerIDNum and CallerIDName headers are also present.
991 Channel Event Logging
992 ---------------------
993 * A new interface, CEL, is introduced here. CEL logs single events, much like
994 the AMI, but it differs from the AMI in that it logs to db backends much
995 like CDR does; is based on the event subsystem introduced by Russell, and
996 can share in all its benefits; allows multiple backends to operate like CDR;
997 is specialized to event data that would be of concern to billing sytems,
998 like CDR. Backends for logging and accounting calls have been produced,
999 but a new CDR backend is still in development.
1003 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
1004 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
1005 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
1006 * Multiple files and formats can now be specified in cdr_custom.conf.
1007 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
1008 See configs/cdr_syslog.conf.sample for more information.
1009 * A 'sequence' field has been added to CDRs which can be combined with
1010 linkedid or uniqueid to uniquely identify a CDR.
1011 * Handling of billsec and duration field has changed. If your table definition
1012 specifies those fields as float,double or similar they will now be logged with
1013 microsecond accuracy instead of a whole integer.
1015 Calendaring for Asterisk
1016 ------------------------
1017 * A new set of modules were added supporing calendar integration with Asterisk.
1018 Dialplan functions for reading from and writing to calendars are included,
1019 as well as the ability to execute dialplan logic upon calendar event notifications.
1020 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
1021 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
1022 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
1023 2003 support does not support forms-based authentication).
1025 Call Completion Supplementary Services for Asterisk
1026 ---------------------------------------------------
1027 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
1028 DAHDI/ISDN supports call completion for the following switch types:
1029 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
1030 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
1032 Multicast RTP Support
1033 ---------------------
1034 * A new RTP engine and channel driver have been added which supports Multicast RTP.
1035 The channel driver can be used with the Page application to perform multicast RTP
1036 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
1037 Type can be either basic or linksys.
1038 Destination is the IP address and port for the RTP packets.
1039 Control address is specific to the linksys type and is used for sending the control
1040 packets unique to them.
1042 Security Events Framework
1043 -------------------------
1044 * Asterisk has a new C API for reporting security events. The module res_security_log
1045 sends these events to the "security" logger level. Currently, AMI is the only
1046 Asterisk component that reports security events. However, SIP support will be
1047 coming soon. For more information on the security events framework, see the
1048 "Asterisk Security Framework" section of the Asterisk wiki at
1049 https://wiki.asterisk.org/wiki/x/wgBQ
1050 * SIP support was added in Asterisk 10
1051 * This API now supports IPv6 addresses
1055 * A technology independent fax frontend (res_fax) has been added to Asterisk.
1056 * A spandsp based fax backend (res_fax_spandsp) has been added.
1057 * The app_fax module has been deprecated in favor of the res_fax module and
1058 the new res_fax_spandsp backend.
1059 * The SendFAX and ReceiveFAX applications now send their log messages to a
1060 'fax' logger level, instead of to the generic logger levels. To see these
1061 messages, the system's logger.conf file will need to direct the 'fax' logger
1062 level to one or more destinations; the logger.conf.sample file includes an
1063 example of how to do this. Note that if the 'fax' logger level is *not*
1064 directed to at least one destination, log messages generated by these
1065 applications will be lost, and that if the 'fax' logger level is directed to
1066 the console, the 'core set verbose' and 'core set debug' CLI commands will
1067 have no effect on whether the messages appear on the console or not.
1071 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
1072 Now, in order to enable transmitting silence during record the transmit_silence
1073 option should be used. transmit_silence_during_record remains a valid option, but
1074 defaults to the behavior of the transmit_silence option.
1075 * Addition of the Unit Test Framework API for managing registration and execution
1076 of unit tests with the purpose of verifying the operation of C functions.
1077 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
1078 XMPP text messages to the remote JID.
1079 * Modules.conf has a new option - "require" - that marks a module as critical for
1080 the execution of Asterisk.
1081 If one of the required modules fail to load, Asterisk will exit with a return
1083 * An 'X' option has been added to the asterisk application which enables #exec support.
1084 This allows #exec to be used in asterisk.conf.
1085 * jabber.conf supports a new option auth_policy that toggles auto user registration.
1086 * A new lockconfdir option has been added to asterisk.conf to protect the
1087 configuration directory (/etc/asterisk by default) during reloads.
1088 * The parkeddynamic option has been added to features.conf to enable the creation
1089 of dynamic parkinglots.
1090 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
1091 the reportalarms config option.
1092 * chan_dahdi supports dialing configuring and dialing by device file name.
1093 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
1094 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
1095 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
1096 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
1097 Handy for the above name-based syntax as it does not depend on
1098 initialization order.
1099 * The Realtime dialplan switch now caches entries for 1 second. This provides a
1100 significant increase in performance (about 3X) for installations using this switchtype.
1101 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
1102 AIS. For more information, please see the Distributed Device State section of the
1103 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1104 * The addition of G.719 pass-through support.
1105 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
1106 during device configuration.
1107 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
1108 have less than 3 lines on the LCD.
1109 * Realtime now supports database failover. See the sample extconfig.conf for details.
1110 * The addition of improved translation path building for wideband codecs. Sample
1111 rate changes during translation are now avoided unless absolutely necessary.
1112 * The addition of the res_stun_monitor module for monitoring and reacting to network
1113 changes while behind a NAT.
1117 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
1118 optionally accept a filename, to apply the setting only to the code generated from
1119 that source file when Asterisk was built. However, there are some modules in Asterisk
1120 that are composed of multiple source files, so this did not result in the behavior
1121 that users expected. In this version, 'core set debug' and 'core set verbose'
1122 can optionally accept *module* names instead (with or without the .so extension),
1123 which applies the setting to the entire module specified, regardless of which source
1124 files it was built from.
1125 * New 'manager show settings' command showing the current settings loaded from
1127 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
1128 the channel hangup request to all channels.
1129 * Added a "core reload" CLI command that executes a global reload of Asterisk.
1131 ------------------------------------------------------------------------------
1132 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
1133 ------------------------------------------------------------------------------
1137 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
1138 Snom phones use this for call pickup of extensions that the phone is
1140 * Added support for setting the domain in the URI for caller of an
1141 outbound call by using the SIPFROMDOMAIN channel variable.
1142 * Added a new configuration option "remotesecret" for authentication to
1143 remote services. For backwards compatibility, "secret" still has the
1144 same function as before, but now you can configure both a remote secret and a
1145 local secret for mutual authentication.
1146 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
1147 the sound will be played to the target of an attended transfer
1148 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
1149 finer control over how many peers Asterisk will qualify and the gap between them
1150 when all peers need to be qualified at the same time.
1151 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
1152 (either globally or for a specific peer), chan_sip will treat any SDP data
1153 it receives as new data and update the media stream accordingly. By
1154 default, Asterisk will only modify the media stream if the SDP session
1155 version received is different from the current SDP session version. This
1156 option is required to interoperate with devices that have non-standard SDP
1157 session version implementations (observed with Microsoft OCS). This option
1158 is disabled by default.
1159 * The parsing of register => lines in sip.conf has been modified to allow a port
1160 to be present in the "user" portion. Please see the sip.conf.sample file for more
1162 * Added support for subscribing to MWI on a remote server and making the status available
1163 as a mailbox. Please see the sip.conf.sample file for more information.
1164 * Added a function to remove SIP headers added in the dialplan before the
1165 first INVITE is generated - SIPRemoveHeader()
1166 * Channel variables set with setvar= in a device configuration is now
1167 set both for inbound and outbound calls.
1168 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
1172 * Added immediate option to iax.conf
1173 * Added forceencryption option to iax.conf
1174 * Added Encryption and Trunk status to manager command "iaxpeers"
1178 * The configuration file now holds separate sections for devices and lines.
1179 Please have a look at configs/skinny.conf.sample and change your skinny.conf
1184 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
1185 support for LibOpenR2. http://www.libopenr2.org/
1186 * The UK option waitfordialtone has been added for use with BT analog
1188 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
1189 is used in conjunction with the 'faxdetect' configuration option. When
1190 'faxbuffers' is used and fax tones are detected, the channel will dynamically
1191 switch to the configured faxbuffers policy. For example, to use 6 buffers
1192 and a 'full' buffer policy for a fax transmission, add:
1194 The faxbuffers configuration will be in affect until the call is torn down.
1195 * Added service message support for 4ESS/5ESS switches.
1199 * For DAHDI channels, the CHANNEL() dialplan function now
1200 supports changing the channel's buffer policy (for the current
1201 call only), using this syntax:
1203 exten => s,n,Set(CHANNEL(buffers)=6,full)
1205 This would change the channel to the 'full' buffer policy and
1206 6 (six) buffers. Possible options for this setting are the same
1207 as those in chan_dahdi.conf.
1208 * Added a new dialplan function, CURLOPT, which permits setting various
1209 options that may be useful with the CURL dialplan function, such as
1210 cookies, proxies, connection timeouts, passwords, etc.
1211 * Permit the syntax and synopsis fields of the corresponding dialplan
1212 functions to be individually set from func_odbc.conf.
1213 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
1214 * func_odbc now may specify an insert query to execute, when the write query
1215 affects 0 rows (usually indicating that no such row exists).
1216 * Added a new dialplan function, LISTFILTER, which permits removing elements
1217 from a set list, by name. Uses the same general syntax as the existing CUT
1218 and FIELDQTY dialplan functions, which also manage lists.
1219 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
1220 obtaining realtime data from the dialplan.
1221 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
1222 a subroutine when using the GoSub() and Return() applications.
1223 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
1224 of "core show function AUDIOHOOK_INHERIT" from the CLI
1225 * Added AES_ENCRYPT. For information on its use, please see the output
1226 of "core show function AES_ENCRYPT" from the CLI
1227 * Added AES_DECRYPT. For information on its use, please see the output
1228 of "core show function AES_DECRYPT" from the CLI
1229 * func_odbc now supports database transactions across multiple queries.
1233 * Scheduled meetme conferences may now have their end times extended by
1235 * app_authenticate now gives the ability to select a prompt other than
1237 * app_directory now pays attention to the searchcontexts setting in
1238 voicemail.conf and will look through all contexts, if no context is
1239 specified in the initial argument.
1240 * A new application, Originate, has been introduced, that allows asynchronous
1241 call origination from the dialplan.
1242 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
1243 in addition to the setting in the "general" context.
1244 * Added ConfBridge dialplan application which does conference bridges without
1245 DAHDI. For information on its use, please see the output of
1246 "core show application ConfBridge" from the CLI.
1250 * The Asterisk CLI has a new command, "channel redirect", which is similar in
1251 operation to the AMI Redirect action.
1252 * extensions.conf now allows you to use keyword "same" to define an extension
1253 without actually specifying an extension. It uses exactly the same pattern
1254 as previously used on the last "exten" line. For example:
1255 exten => 123,1,NoOp(something)
1256 same => n,SomethingElse()
1257 * musiconhold.conf classes of type 'files' can now use relative directory paths,
1258 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
1259 * All deprecated CLI commands are removed from the sourcecode. They are now handled
1260 by the new clialiases module. See cli_aliases.conf.sample file.
1261 * Times within timespecs are now accurate down to the minute. This is a change
1262 from historical Asterisk, which only provided timespecs rounded to the nearest
1263 even (read: evenly divisible by 2) minute mark.
1264 * The realtime switch now supports an option flag, 'p', which disables searches for
1266 * In addition to a time range and date range, timespecs now accept a 5th optional
1267 argument, timezone. This allows you to perform time checks on alternate
1268 timezones, especially if those daylight savings time ranges vary from your
1269 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
1271 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
1272 give you the correct output for an asterisk box behind nat. It will give you the
1273 externhost and localnet settings.
1274 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
1275 can connect calls in passthrough mode, as well as record and play back files.
1276 * Successful and unsuccessful call pickup can now be alerted through sounds, by
1277 using pickupsound and pickupfailsound in features.conf.
1278 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
1279 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
1280 instead of the /var/run/asterisk.pid where it used to be. This will make
1281 installs as non-root easier to manage.
1286 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
1287 be written; they will no longer be explicitly written.
1289 Asterisk Manager Interface
1290 --------------------------
1291 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
1292 a non-empty value) in your request. If you do this, any pending AMI events will
1293 *not* be included in the response to your request as they would normally, but
1294 will be left in the event queue for the next request you make to retrieve. For
1295 some applications, this will allow you to guarantee that you will only see
1296 events in responses to 'WaitEvent' actions, and can better know when to expect them.
1297 To know whether the Asterisk server supports this header or not, your client can
1298 inspect the first response back from the server to see if it includes this header:
1300 Pragma: SuppressEvents
1302 If this is included, the server supports event suppression.
1304 * Added 4 new Actions to list skinny device(s) and line(s)
1310 LDAP Schema File Additions
1311 --------------------------
1312 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
1313 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
1315 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
1316 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
1317 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
1318 * Removed redundant IPaddr (there's already IPAddress)
1319 - Gives more configuration Flags for SIP-Users available (tested)
1320 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
1321 without extensibleObject (which really should be the last resort); gives
1322 also additional possibilities for LDAP-filter
1324 ------------------------------------------------------------------------------
1325 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
1326 ------------------------------------------------------------------------------
1328 Device State Handling
1329 ---------------------
1330 * The event infrastructure in Asterisk got another big update to help support
1331 distributed events. It currently supports distributed device state and
1332 distributed Voicemail MWI (Message Waiting Indication). A new module has
1333 been merged, res_ais, which facilitates communicating events between servers.
1334 It uses the SAForum AIS (Service Availability Forum Application Interface
1335 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
1336 a cluster of Asterisk servers, and to share events between them. For more
1337 information on setting this up, refer to the Distributed Device State section
1338 of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1342 * Added a new dialplan function, AST_CONFIG(), which allows you to access
1343 variables from an Asterisk configuration file.
1344 * The JACK_HOOK function now has a c() option to supply a custom client name.
1345 * Added two new dialplan functions from libspeex for audio gain control and
1346 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
1347 rx directions of a channel from the dialplan.
1348 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
1349 based on other parameters. The default is still to search based on the
1350 forwarding station ID. However, there are new options that allow you to search
1351 based on the message desk terminal ID, or the message desk number.
1352 * TIMEOUT() has been modified to be accurate down to the millisecond.
1353 * ENUM*() functions now include the following new options:
1354 - 'u' returns the full URI and does not strip off the URI-scheme.
1355 - 's' triggers ISN specific rewriting
1356 - 'i' looks for branches into an Infrastructure ENUM tree
1357 - 'd' for a direct DNS lookup without any flipping of digits.
1358 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
1359 * CHANNEL() now has options for the maximum, minimum, and standard or normal
1360 deviation of jitter, rtt, and loss for a call using chan_sip.
1362 DAHDI channel driver (chan_dahdi) Changes
1363 ----------------------------------------
1364 * Channels can now be configured using named sections in chan_dahdi.conf, just
1365 like other channel drivers, including the use of templates.
1366 * The default for pridialplan has changed from 'national' to 'unknown'.
1370 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
1371 to something that matches the pattern a hint will be created using the contents
1372 and variables evaluated.
1373 * Dialplan matching has been extended to allow an extension to return to the
1374 PBX core to wait for more digits. This is done by using the new dialplan
1375 application called "Incomplete". This will permit a whole new level of
1376 extension control, by giving the administrator more control over early
1377 matches employing one of the short-circuit pattern match operators. Note
1378 that custom applications can trigger this same behavior by returning the
1379 special value AST_PBX_INCOMPLETE.
1383 * Directory now permits both first and last names to be matched at the same
1384 time. In addition, the number of digits to enter of the name can be set in
1385 the arguments to Directory; previously, you could enter only 3, regardless
1386 of how many names are in your company. For large companies, this should be
1388 * Voicemail now permits a mailbox setting to wrap around from first to last
1389 messages, if the "messagewrap" option is set to a true value.
1390 * Voicemail now permits an external script to be run, for password validation.
1391 The script should output "VALID" or "INVALID" on stdout, depending upon the
1392 wish to validate or invalidate the password given. Arguments are:
1393 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
1395 * Dial has a new option: F(context^extension^pri), which permits a callee to
1396 continue in the dialplan, at the specified label, if the caller hangs up.
1397 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
1398 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
1399 * The Jack application now has a c() option to supply a custom client name.
1400 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
1401 like the pre-existing whisper mode, except that the spy can also talk to the
1402 participant on the bridged channel as well.
1403 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
1404 to be spoken instead of the channel name or number. For more information on the
1405 use of this option, issue the command "core show application ChanSpy" from the
1407 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
1408 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
1409 words, if using the 'd' option, it is not possible to enter a number to append to
1410 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
1411 change to whisper mode, and pressing 6 will change to barge mode.
1412 * ExternalIVR now takes several options that affect the way it performs, as
1413 well as having several new commands. Please see the External IVR page on the Asterisk
1414 wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
1415 * Added ability to communicate over a TCP socket instead of forking a child process for the
1416 ExternalIVR application.
1417 * ChanIsAvail has a new option, 'a', which will return all available channels instead
1418 of just the first one if you give the function more then one channel to check.
1419 * PrivacyManager now takes an option where you can specify a context where the
1420 given number will be matched. This way you have more control over who is allowed
1421 and it stops the people who blindly enter 10 digits.
1422 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
1423 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
1424 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
1425 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
1426 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
1427 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
1428 * The Dial() application no longer copies the language used by the caller to the callee's
1429 channel. If you desire for the caller's channel's language to be used for file playback
1430 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
1431 * SendImage() no longer hangs up the channel on error; instead, it sets the
1432 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
1433 'UNSUPPORTED'. This change makes SendImage() more consistent with other
1435 * Park has a new option, 's', which silences the announcement of the parking space number.
1436 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
1437 invalid input and will be assumed to mean that no timeout is desired.
1441 * Added DNS manager support to registrations for peers referencing peer entries.
1442 DNS manager runs in the background which allows DNS lookups to be run asynchronously
1443 as well as periodically updating the IP address. These properties allow for
1444 better performance as well as recovery in the event of an IP change.
1445 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
1446 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
1447 These changes also provide performance improvements for call setup and tear down.
1448 * Added ability to specify registration expiry time on a per registration basis in
1450 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
1452 * Added t38pt_usertpsource option. See sip.conf.sample for details.
1453 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
1454 * 'sip show peers' and 'sip show users' display their entries sorted in
1455 alphabetical order, as opposed to the order they were in, in the config
1457 * Videosupport now supports an additional option, "always", which always sets
1458 up video RTP ports, even on clients that don't support it. This helps with
1459 callfiles and certain transfers to ensure that if two video phones are
1460 connected, they will always share video feeds.
1464 * Existing DNS manager lookups extended to check for SRV records.
1465 * IAX2 encryption support has been improved to support periodic key rotation
1466 within a call for enhanced security. The option "keyrotate" has been
1467 provided to disable this functionality to preserve backwards compatibility
1468 with older versions of IAX2 that do not support key rotation.
1472 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
1473 data tree based on the given <path>.
1474 * New CLI command "data show providers" that will display all the registered
1476 * New CLI command, "config reload <file.conf>" which reloads any module that
1477 references that particular configuration file. Also added "config list"
1478 which shows which configuration files are in use.
1479 * New CLI commands, "pri show version" and "ss7 show version" that will
1480 display which version of libpri and libss7 are being used, respectively.
1481 A new API call was added so trunk will now have to be compiled against
1482 a versions of libpri and libss7 that have them or it will not know that
1483 these libraries exist.
1484 * The commands "core show globals", "core set global" and "core set chanvar" has
1485 been deprecated in favor of the more semanticly correct "dialplan show globals",
1486 "dialplan set chanvar" and "dialplan set global".
1487 * New CLI command "dialplan show chanvar" to list all variables associated
1488 with a given channel.
1492 * Addresses managed by DNS manager now can check to see if there is a DNS
1493 SRV record for a given domain and will use that hostname/port if present.
1495 AMI - The manager (TCP/TLS/HTTP)
1496 --------------------------------
1497 * The Status command now takes an optional list of variables to display
1498 along with channel status.
1499 * The QueueEntry event now also includes the channel's uniqueid
1503 * res_odbc no longer has a limit of 1023 total possible unshared connections,
1504 as some people were running into this limit. This limit has been increased
1509 * The TRANSFER queue log entry now includes the the caller's original
1510 position in the transferred-from queue.
1511 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
1512 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
1513 as well as an explanation about timeout options in general
1514 * Added a new option - C - for forcing the "answered elsewhere" flag on
1515 cancellation of calls in to members of the queue. This is to avoid the
1516 call to a member of a queue having the call listed as a "missed call".
1520 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
1521 adaptive capabilities. What this means in practical terms is that if your
1522 realtime table lacks critical fields, Asterisk will now emit warnings to
1523 that effect. Also, some of the realtime drivers have the ability (if
1524 configured) to automatically add those columns to the table with the
1525 correct type and length.
1529 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
1530 the 'setvar' option to cause a given audio file to be played upon completion
1531 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
1532 Skinny channels only.
1533 * You can now compile Asterisk against the Hoard Memory Allocator, see the
1534 Hoard page on the Asterisk wiki for more information:
1535 https://wiki.asterisk.org/wiki/x/pQBB
1536 * Config file variables may now be appended to, by using the '+=' append
1537 operator. This is most helpful when working with long SQL queries in
1538 func_odbc.conf, as the queries no longer need to be specified on a single
1540 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
1541 which will add a second to the billsec when the ending
1542 time is set, if the number in the microseconds field of the end time is
1543 greater than the number of microseconds in the answer time. This allows
1544 users to count the 'initiated' seconds in their billing records.
1546 ------------------------------------------------------------------------------
1547 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
1548 ------------------------------------------------------------------------------
1550 AMI - The manager (TCP/TLS/HTTP)
1551 --------------------------------
1552 * Manager has undergone a lot of changes, all of them documented
1553 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
1554 * Manager version has changed to 1.1
1555 * Added a new action 'CoreShowChannels' to list currently defined channels
1556 and some information about them.
1557 * Added a new action 'SIPshowregistry' to list SIP registrations.
1558 * Added TLS support for the manager interface and HTTP server
1559 * Added the URI redirect option for the built-in HTTP server
1560 * The output of CallerID in Manager events is now more consistent.
1561 CallerIDNum is used for number and CallerIDName for name.
1562 * Enable https support for builtin web server.
1563 See configs/http.conf.sample for details.
1564 * Added a new action, GetConfigJSON, which can return the contents of an
1565 Asterisk configuration file in JSON format. This is intended to help
1566 improve the performance of AJAX applications using the manager interface
1568 * SIP and IAX manager events now use "ChannelType" in all cases where we
1569 indicate channel driver. Previously, we used a mixture of "Channel"
1570 and "ChannelDriver" headers.
1571 * Added a "Bridge" action which allows you to bridge any two channels that
1572 are currently active on the system.
1573 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
1574 the voicemail users setup.
1575 * Added 'DBDel' and 'DBDelTree' manager commands.
1576 * cdr_manager now reports events via the "cdr" level, separating it from
1577 the very verbose "call" level.
1578 * Manager users are now stored in memory. If you change the manager account
1579 list (delete or add accounts) you need to reload manager.
1580 * Added Masquerade manager event for when a masquerade happens between
1582 * Added "manager reload" command for the CLI
1583 * Lots of commands that only provided information are now allowed under the
1584 Reporting privilege, instead of only under Call or System.
1585 * The IAX* commands now require either System or Reporting privilege, to
1586 mirror the privileges of the SIP* commands.
1587 * Added ability to retrieve list of categories in a config file.
1588 * Added ability to retrieve the content of a particular category.
1589 * Added ability to empty a context.
1590 * Created new action to create a new file.
1591 * Updated delete action to allow deletion by line number with respect to category.
1592 * Added new action insert to add new variable to category at specified line.
1593 * Updated action newcat to allow new category to be inserted in file above another
1595 * Added new event "JitterBufStats" in the IAX2 channel
1596 * Originate now requires the Originate privilege and, if you want to call out
1597 to a subshell, it requires the System privilege, as well. This was done to
1598 enhance manager security.
1599 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
1600 * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
1601 or manager show command Atxfer from the CLI
1602 * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
1603 details or manager show command IAXregistry from the CLI
1607 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
1608 state in the dialplan, as well as creating custom device states that are
1609 controllable from the dialplan.
1610 * Extend CALLERID() function with "pres" and "ton" parameters to
1611 fetch string representation of calling number presentation indicator
1612 and numeric representation of type of calling number value.
1613 * MailboxExists converted to dialplan function
1614 * A new option to Dial() for telling IP phones not to count the call
1615 as "missed" when dial times out and cancels.
1616 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
1617 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
1618 held for any given channel. Also, locks are automatically freed when a
1620 * Added HINT() dialplan function that allows retrieving hint information.
1621 Hints are mappings between extensions and devices for the sake of
1622 determining the state of an extension. This function can retrieve the list
1623 of devices or the name associated with a hint.
1624 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
1626 * Added SYSINFO() dialplan function which allows retrieval of system information
1627 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
1628 the existence of a dialplan target.
1629 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
1630 upper and lower case, respectively.
1631 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
1632 ID for the call (not the Asterisk call ID or unique ID), provided that the
1633 channel driver supports this. For SIP, you get the SIP call-ID for the
1634 bridged channel which you can store in the CDR with a custom field.
1638 * Added CLI permissions, config file: cli_permissions.conf
1639 default is to allow all commands for every local user/group.
1640 Also this new feature added three new CLI commands:
1641 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
1642 - cli reload permissions
1643 - cli show permissions
1644 * New CLI command "core show hint" (usage: core show hint <exten>)
1645 * New CLI command "core show settings"
1646 * Added 'core show channels count' CLI command.
1647 * Added the ability to set the core debug and verbose values on a per-file basis.
1648 * Added 'queue pause member' and 'queue unpause member' CLI commands
1649 * Ability to set process limits ("ulimit") without restarting Asterisk
1650 * Enhanced "agi debug" to print the channel name as a prefix to the debug
1651 output to make debugging on busy systems much easier.
1652 * New CLI commands "dialplan set extenpatternmatching true/false"
1653 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
1654 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
1655 listed in the startup_commands section of cli.conf will get executed.
1656 * Added a CLI command, "devstate change", which allows you to set custom device
1657 states from the func_devstate module that provides the DEVICE_STATE() function
1658 and handling of the "Custom:" devices.
1659 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
1660 sorted into the different possible callbacks, with the number of entries
1661 currently scheduled for each. Gives you a feel for how busy the sip channel
1663 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
1664 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
1665 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
1669 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
1670 option is enabled, Asterisk will watch for a CNG tone in the incoming audio
1671 for a received call. If it is detected, the channel will jump to the
1672 'fax' extension in the dialplan.
1673 * The default SIP useragent= identifier now includes the Asterisk version
1674 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
1675 If set, and the incoming request carries authentication info,
1676 the username to match in the users list is taken from the Digest header
1677 rather than from the From: field. This feature is considered experimental.
1678 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
1679 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
1680 * The "localmask" setting was removed in version 1.2 and the reminder about it
1681 being removed is now also removed.
1682 * A new option "busylevel" for setting a level of calls where asterisk reports
1683 a device as busy, to separate it from call-limit. This value is also added
1684 to the SIP_PEER dialplan function.
1685 * A new realtime family called "sipregs" is now supported to store SIP registration
1686 data. If this family is defined, "sippeers" will be used for configuration and
1687 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
1688 registration data, as before.
1689 * The SIPPEER function have new options for port address, call and pickup groups
1690 * Added support for T.140 realtime text in SIP/RTP
1691 * The "checkmwi" option has been removed from sip.conf, as it is no longer
1692 required due to the restructuring of how MWI is handled. See the descriptions
1693 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
1694 for more information.
1695 * Added rtpdest option to CHANNEL() dialplan function.
1696 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
1697 * SIP now adds a header to the CANCEL if the call was answered by another phone
1698 in the same dial command, or if the new c option in dial() is used.
1699 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
1700 states it is not needed. For phones, however, that do require it the "registertrying" option
1701 has been added so it can be enabled.
1702 * A new option called "callcounter" (global/peer/user level) enables call counters needed
1703 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
1704 used to enable this functionality).
1705 * New settings for timer T1 and timer B on a global level or per device. This makes it
1706 possible to force timeout faster on non-responsive SIP servers. These settings are
1707 considered advanced, so don't use them unless you have a problem.
1708 * Added a dial string option to be able to set the To: header in an INVITE to any
1710 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
1711 the qualify frequency.
1712 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
1713 were not properly torn down due to network or endpoint failures during an established
1715 * Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB
1716 and configs/sip.conf.sample for more information on how it is used.
1717 * Added a new configuration option "authfailureevents" that enables manager events when
1718 a peer can't authenticate properly.
1719 * Added DNS manager support to registrations for peers not referencing a peer entry.
1723 * Added the trunkmaxsize configuration option to chan_iax2.
1724 * Added the srvlookup option to iax.conf
1725 * Added support for OSP. The token is set and retrieved through the CHANNEL()
1728 XMPP Google Talk/Jingle changes
1729 -------------------------------
1730 * Added the bindaddr option to gtalk.conf.
1734 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
1735 * Proper codec support in chan_skinny.
1736 * Added settings for IP and Ethernet QoS requests
1740 * Added separate settings for media QoS in mgcp.conf
1742 Console Channel Driver changes
1743 ------------------------------
1744 * Added experimental support for video send & receive to chan_oss.
1745 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
1748 Phone channel changes (chan_phone)
1749 ----------------------------------
1750 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
1752 H.323 channel Changes
1753 ---------------------
1754 * H323 remote hold notification support added (by NOTIFY message
1755 and/or H.450 supplementary service)
1757 Local channel changes
1758 ---------------------
1759 * The device state functionality in the Local channel driver has been updated
1760 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
1761 to just UNKNOWN if the extension exists.
1762 * Added jitterbuffer support for chan_local. This allows you to use the
1763 generic jitterbuffer on incoming calls going to Asterisk applications.
1764 For example, this would allow you to use a jitterbuffer for an incoming
1765 SIP call to Voicemail by putting a Local channel in the middle. This
1766 feature is enabled by using the 'j' option in the Dial string to the Local
1767 channel in conjunction with the existing 'n' option for local channels.
1768 * A 'b' option has been added which causes chan_local to return the actual channel
1769 that is behind it when queried. This is useful for transfer scenarios as the
1770 actual channel will be transferred, not the Local channel.
1772 Agent channel changes
1773 ----------------------
1774 * The ackcall and endcall options are now supplemented with options acceptdtmf
1775 and enddtmf. These allow for the DTMF keypress to be configurable. The options
1776 default to their old hard-coded values ('#' and '*' respectively) so this should
1777 not break any existing agent installations.
1779 DAHDI channel driver (chan_dahdi) Changes
1780 ----------------------------------------
1781 * SS7 support (via libss7 library)
1782 * In India, some carriers transmit CID via dtmf. Some code has been added
1783 that will handle some situations. The cidstart=polarity_IN choice has been added for
1784 those carriers that transmit CID via dtmf after a polarity change.
1785 * CID matching information is now shown when doing 'dialplan show'.
1786 * Added dahdi show version CLI command.
1787 * Added setvar support to chan_dahdi.conf channel entries.
1788 * Added two new options: mwimonitor and mwimonitornotify. These options allow
1789 you to enable MWI monitoring on FXO lines. When the MWI state changes,
1790 the script specified in the mwimonitornotify option is executed. An internal
1791 event indicating the new state of the mailbox is also generated, so that
1792 the normal MWI facilities in Asterisk work as usual.
1793 * Added signalling type 'auto', which attempts to use the same signalling type
1794 for a channel as configured in DAHDI. This is primarily designed for analog
1795 ports, but will also work for digital ports that are configured for FXS or FXO
1796 signalling types. This mode is also the default now, so if your chan_dahdi.conf
1797 does not specify signalling for a channel (which is unlikely as the sample
1798 configuration file has always recommended specifying it for every channel) then
1799 the 'auto' mode will be used for that channel if possible.
1800 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
1801 state for a channel; also ensured that the DNDState Manager event is
1802 emitted no matter how the DND state is set or cleared.
1806 * Added a new channel driver, chan_unistim. See the Asterisk wiki at
1807 https://wiki.asterisk.org/wiki/x/vgsiAQ and configs/unistim.conf.sample
1808 for details. This new channel driver allows you to use Nortel i2002,
1809 i2004, and i2050 phones with Asterisk.
1810 * Added a new channel driver, chan_console, which uses portaudio as a cross
1811 platform audio interface. It was written as a channel driver that would
1812 work with Mac CoreAudio, but portaudio supports a number of other audio
1813 interfaces, as well. Note that this channel driver requires v19 or higher
1814 of portaudio; older versions have a different API.
1818 * Added the ability to specify arguments to the Dial application when using
1819 the DUNDi switch in the dialplan.
1820 * Added the ability to set weights for responses dynamically. This can be
1821 done using a global variable or a dialplan function. Using the SHELL()
1822 function would allow you to have an external script set the weight for
1824 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
1825 functions will allow you to initiate a DUNDi query from the dialplan,
1826 find out how many results there are, and access each one.
1827 * Added the ability to specifiy a port for a dundi peer.
1831 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
1832 functions will allow you to initiate an ENUM lookup from the dialplan,
1833 and Asterisk will cache the results. ENUMRESULT can be used to access
1834 the results without doing multiple DNS queries.
1838 * Added the ability to customize which sound files are used for some of the
1839 prompts within the Voicemail application by changing them in voicemail.conf
1840 * Added the ability for the "voicemail show users" CLI command to show users
1841 configured by the dynamic realtime configuration method.
1842 * MWI (Message Waiting Indication) handling has been significantly
1843 restructured internally to Asterisk. It is now totally event based
1844 instead of polling based. The voicemail application will notify other
1845 modules that have subscribed to MWI events when something in the mailbox
1847 This also means that if any other entity outside of Asterisk is changing
1848 the contents of mailboxes, then the voicemail application still needs to
1849 poll for changes. Examples of situations that would require this option
1850 are web interfaces to voicemail or an email client in the case of using
1851 IMAP storage. So, two new options have been added to voicemail.conf
1852 to account for this: "pollmailboxes" and "pollfreq". See the sample
1853 configuration file for details.
1854 * Added "tw" language support
1855 * Added support for storage of greetings using an IMAP server
1856 * Added ability to customize forward, reverse, stop, and pause keys for message playback
1857 * SMDI is now enabled in voicemail using the smdienable option.
1858 * A "lockmode" option has been added to asterisk.conf to configure the file
1859 locking method used for voicemail, and potentially other things in the
1860 future. The default is the old behavior, lockfile. However, there is a
1861 new method, "flock", that uses a different method for situations where the
1862 lockfile will not work, such as on SMB/CIFS mounts.
1863 * Added the ability to backup deleted messages, to ease recovery in the case
1864 that a user accidentally deletes a message, and discovers that they need it.
1865 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
1866 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
1867 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
1868 voicemail boxes. The SMDI interface can also poll for MWI changes when some
1869 outside entity is modifying the state of the mailbox (such as IMAP storage or
1870 a web interface of some kind).
1871 * Added the support for marking messages as "urgent." There are two methods to accomplish
1872 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
1873 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
1874 the message as urgent after he has recorded a voicemail by following the voice instructions.
1875 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
1880 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
1881 used across multiple queues.
1882 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
1883 setqueueentryvar options for each queue, see queues.conf.sample for details.
1884 * Added keepstats option to queues.conf which will keep queue
1885 statistics during a reload.
1886 * setinterfacevar option in queues.conf also now sets a variable
1887 called MEMBERNAME which contains the member's name.
1888 * Added 'Strategy' field to manager event QueueParams which represents
1889 the queue strategy in use.
1890 * Added option to run macro when a queue member is connected to a caller,
1891 see queues.conf.sample for details.
1892 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
1893 does not count paused queue members as unavailable.
1894 * Added min-announce-frequency option to queues.conf which allows you to control the
1895 minimum amount of time between queue announcements for use when the caller's queue
1896 position changes frequently.
1897 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
1899 * Added ability for non-realtime queues to have realtime members
1900 * Added the "linear" strategy to queues.
1901 * Added the "wrandom" strategy to queues.
1902 * Added new channel variable QUEUE_MIN_PENALTY
1903 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
1904 rules in queuerules.conf. See configs/queuerules.conf.sample for details
1905 * Added a new parameter for member definition, called state_interface. This may be
1906 used so that a member may be called via one interface but have a different interface's
1907 device state reported.
1908 * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
1909 "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
1910 "manager show command QueueReset."
1911 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
1912 specified by the periodic-announce option, then one will be chosen randomly when it is time
1913 to play a periodic announcment
1914 * New configuration options: announce-position now takes two more values in addition to "yes" and
1915 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
1916 announce-position-limit. By setting announce-position to "limit" callers will only have their
1917 position announced if their position is less than what is specified by announce-position-limit.
1918 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
1919 will be told that their are more than announce-position-limit callers waiting.
1920 * Two new queue log events have been added. An ADDMEMBER event will be logged
1921 when a realtime queue member is added and a REMOVEMEMBER event will be logged
1922 when a realtime queue member is removed. Since there is no calling channel associated
1923 with these events, the string "REALTIME" is placed where the channel's unique id
1924 is typically placed.
1925 * The configuration method for the "joinempty" and "leavewhenempty" options has
1926 changed to a comma-separated list of methods of determining member availability
1927 instead of vague terms such as "yes," "loose," "no," and "strict." These old four
1928 values are still accepted for backwards-compatibility, though.
1929 * The average talktime is now calculated on queues. This information is reported via the
1930 CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
1931 and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
1936 * The 'o' option to provide an optimization has been removed and its functionality
1937 has been enabled by default.
1938 * When a conference is created, the UNIQUEID of the channel that caused it to be
1939 created is stored. Then, every channel that joins the conference will have the
1940 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
1941 callers that come and go from long standing conferences.
1942 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
1943 except it does operations on a channel by name, instead of number in a conference.
1944 This is a very useful feature in combination with the 'X' option to ChanSpy.
1945 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
1947 * Added new RealTime functionality to provide support for scheduled conferencing.
1948 This includes optional messages to the caller if they attempt to join before
1949 the schedule start time, or to allow the caller to join the conference early.
1950 Also included is optional support for limiting the number of callers per
1951 RealTime conference.
1952 * Added the S() and L() options to the MeetMe application. These are pretty
1953 much identical to the S() and L() options to Dial(). They let you set
1954 timeouts for the conference, as well as have warning sounds played to
1955 let the caller know how much time is left, and when it is running out.
1956 * Added the ability to do "meetme concise" with the "meetme" CLI command.
1957 This extends the concise capabilities of this CLI command to include
1958 listing all conferences, instead of an addition to the other sub commands
1959 for the "meetme" command.
1960 * Added the ability to specify the music on hold class used to play into the
1961 conference when there is only one member and the M option is used.
1962 * Added MEETME_INFO dialplan function which provides a way to query
1963 various properties of a Meetme conference.
1964 * Added new admin features: *81: Roll call, *82: eject all, *83: mute all,
1965 and *84: record in-conf
1967 Other Dialplan Application Changes
1968 ----------------------------------
1969 * Argument support for Gosub application
1970 * From the to-do lists: straighten out the app timeout args:
1971 Wait() app now really does 0.3 seconds- was truncating arg to an int.
1972 WaitExten() same as Wait().
1973 Congestion() - Now takes floating pt. argument.
1974 Busy() - now takes floating pt. argument.
1975 Read() - timeout now can be floating pt.
1976 WaitForRing() now takes floating pt timeout arg.
1977 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
1978 * Added 's' option to Page application.
1979 * Added an optional timeout argument to the Page application.
1980 * Added 'E', 'V', and 'P' commands to ExternalIVR.
1981 * Added 'o' and 'X' options to Chanspy.
1982 * Added a new dialplan application, Bridge, which allows you to bridge the
1983 calling channel to any other active channel on the system.
1984 * Added the ability to specify a music on hold class to play instead of ringing
1985 for the SLATrunk application.
1986 * The Read application no longer exits the dialplan on error. Instead, it sets
1987 READSTATUS to ERROR, which you can catch and handle separately.
1988 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
1989 of asking for verification of each name, one at a time.
1990 * Privacy() no longer uses privacy.conf, as all options are specifyable as
1991 direct options to the app.
1992 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
1994 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
1995 * The ChannelRedirect application no longer exits the dialplan if the given channel
1996 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
1997 or NOCHANNEL if the given channel was not found.
1998 * The silencethreshold setting that was previously configurable in multiple
1999 applications is now settable globally via dsp.conf.
2001 Music On Hold Changes
2002 ---------------------
2003 * A new option, "digit", has been added for music on hold classes in
2004 musiconhold.conf. If this is set for a music on hold class, a caller
2005 listening to music on hold can press this digit to switch to listening
2006 to this music on hold class.
2007 * Support for realtime music on hold has been added.
2008 * In conjunction with the realtime music on hold, a general section has
2009 been added to musiconhold.conf, its sole variable is cachertclasses. If this
2010 is set, then music on hold classes found in realtime will be cached in memory.
2014 * AEL upgraded to use the Gosub with Arguments instead
2015 of Macro application, to hopefully reduce the problems
2016 seen with the artificially low stack ceiling that
2017 Macro bumps into. Macros can only call other Macros
2018 to a depth of 7. Tests run using gosub, show depths
2019 limited only by virtual memory. A small test demonstrated
2020 recursive call depths of 100,000 without problems.
2021 -- in addition to this, all apps that allowed a macro
2022 to be called, as in Dial, queues, etc, are now allowing
2023 a gosub call in similar fashion.
2024 * AEL now generates LOCAL(argname) declarations when it
2025 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
2026 etc. That makes the arguments local in scope. The user
2027 can define their own local variables in macros, now,
2028 by saying "local myvar=someval;" or using Set() in this
2029 fashion: Set(LOCAL(myvar)=someval); ("local" is now
2031 * utils/conf2ael introduced. Will convert an extensions.conf
2032 file into extensions.ael. Very crude and unfinished, but
2033 will be improved as time goes by. Should be useful for a
2034 first pass at conversion.
2035 * aelparse will now read extensions.conf to see if a referenced
2036 macro or context is there before issueing a warning.
2037 * AEL parser sets a local channel variable ~~EXTEN~~, to
2038 preserve the value of ${EXTEN} thru switch statements.
2039 * New operator in $[...] expressions: the ~~ operator serves
2040 as a concatenation operator. AT THE MOMENT, it is really only
2041 necessary and useful in AEL, especially in if() expressions.
2042 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
2043 any enclosing double-quotes, and evaluate to the value of a
2044 concatenated with the value of b. For example if a is set to
2045 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
2046 evaluate to xyzabc .
2049 Call Features (res_features) Changes
2050 ------------------------------------
2051 * Added the parkedcalltransfers option to features.conf
2052 * Added parkedcallparking option to control one touch parking w/ parking
2054 * Added parkedcallhangup option to control disconnect feature w/ parking
2056 * Added parkedcallrecording option to control one-touch record w/ parking
2058 * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
2059 parkedcalltransfers option support for multiple parking lots.
2060 * Added BRIDGE_FEATURES variable to set available features for a channel
2061 * The built-in method for doing attended transfers has been updated to
2062 include some new options that allow you to have the transferee sent
2063 back to the person that did the transfer if the transfer is not successful.
2064 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
2065 in features.conf.sample.
2066 * Added support for configuring named groups of custom call features in
2067 features.conf. This means that features can be written a single time, and
2068 then mapped into groups of features for different key mappings or easier
2070 * Updated the ParkedCall application to allow you to not specify a parking
2071 extension. If you don't specify a parking space to pick up, it will grab
2072 the first one available.
2073 * Added cli command 'features reload' to reload call features from features.conf
2074 * Moved into core asterisk binary.
2075 * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
2076 * Added the ability for custom parking lots to be configured with their own
2077 parking extension with the parkext option.
2079 Language Support Changes
2080 ------------------------
2081 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
2082 * Added support for the Hungarian language for saying numbers, dates, and times.
2086 * Added SPEECH commands for speech recognition. A complete listing can be found
2088 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
2089 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
2090 does not behave as expected; the native command needs to be used, instead.
2091 * Added the ability to perform SRV lookups on fast AGI calls. To use this
2092 feature, simply use hagi: instead of agi: as the protocol portion
2093 of the URI parameter to the AGI function call in your dial plan. Also note
2094 that specifying a port number in the AGI URI will disable SRV lookups,
2095 even if you use the hagi: protocol.
2096 * No longer support MSG_OOB flag on HANGUP.
2100 * Added rotatestrategy option to logger.conf, along with two new options:
2101 "timestamp" which will use the time to name the logger files instead of
2102 sequence number; and "rotate", which rotates the names of the log files,
2103 similar to the way syslog rotates files.
2104 * Added exec_after_rotate option to logger.conf, which allows a system
2105 command to be run after rotation. This is primarily useful with
2106 rotatestrategy=rotate, to allow a limit on the number of log files kept
2107 and to ensure that the oldest log file gets deleted.
2108 * Added realtime support for the queue log
2112 * The cdr_manager module has a [mappings] feature, like cdr_custom,
2113 to add fields to the manager event from the CDR variables.
2114 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
2115 backend database CDR table. Specifically, additional, non-standard
2116 columns are supported, merely by setting the corresponding CDR variable in
2117 your dialplan. In addition, you may alias any column to another name (for
2118 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
2119 simply "alias src => ANI" in the configuration file). Records may be
2120 posted to more than one backend, simply by specifying multiple categories
2121 in the configuration file. And finally, you may filter which CDRs get
2122 posted to each backend, by specifying a filter (which the record must
2123 match) for the particular category. Filters are additive (meaning all
2124 rules must match to post that CDR).
2125 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
2126 module. Specifically, you may add additional columns into the table and
2127 they will be set, if you set the corresponding CDR variable name. Also,
2128 if you omit columns in your database table, they will be silently skipped
2129 (but a record will still be inserted, based on what columns remain). Note
2130 that the other two features from cdr_adaptive_odbc (alias and filter) are
2131 not currently supported.
2132 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
2133 has been disabled using the NoCDR application.
2135 Miscellaneous New Modules
2136 -------------------------
2137 * Added a new CDR module, cdr_sqlite3_custom.
2138 * Added a new realtime configuration module, res_config_sqlite
2139 * Added a new codec translation module, codec_resample, which re-samples
2140 signed linear audio between 8 kHz and 16 kHz to help support wideband
2142 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
2143 based on configuration templates that use Asterisk dialplan function and
2144 variable substitution. It should be possible to create phone profiles and
2145 templates that work for the majority of phones provisioned over http. It
2146 is currently only intended to provision a single user account per phone.
2147 An example profile and set of templates for Polycom phones is provided.
2148 NOTE: Polycom firmware is not included, but should be placed in
2149 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
2150 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
2151 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
2152 provided; there is a JACK() application, and a JACK_HOOK() function. Both
2153 interfaces create an input and output JACK port. The application makes
2154 these ports the endpoint of the call. The audio coming from the channel
2155 goes out the output port and whatever comes back in on the input port is
2156 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
2157 audiohook on the channel. This lets you run the audio coming from a
2158 channel through JACK, and whatever comes back in is what gets forwarded
2159 on as the channel's audio. This is very useful for building custom
2160 vocoders or doing recording or analysis of the channel's audio in another
2162 * Added a new module, res_config_curl, which permits using a HTTP POST url
2163 to retrieve, create, update, and delete realtime information from a remote
2164 web server. Note that this module requires func_curl.so to be loaded for
2165 backend functionality.
2166 * Added a new module, res_config_ldap, which permits the use of an LDAP
2167 server for realtime data access.
2168 * Added support for writing and running your dialplan in lua using the pbx_lua
2169 module. See configs/extensions.lua.sample for examples of how to do this.
2173 * Ability to use libcap to set high ToS bits when non-root
2174 on Linux. If configure is unable to find libcap then you
2175 can use --with-cap to specify the path.
2176 * Added maxfiles option to options section of asterisk.conf which allows you to specify
2177 what Asterisk should set as the maximum number of open files when it loads.
2178 * Added the jittertargetextra configuration option.
2179 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
2180 configuration files for the IP channel drivers. The new option is "cos".
2181 This information is also documented on the Asterisk wiki at
2182 https://wiki.asterisk.org/wiki/x/EYBG
2183 * When originating a call using AMI or pbx_spool that fails the reason for failure
2184 will now be available in the failed extension using the REASON dialplan variable.
2185 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
2186 It allows you to configure a prefix for auto-monitor recordings.
2187 * A new extension pattern matching algorithm, based on a trie, is introduced
2188 here, that could noticeably speed up mid-sized to large dialplans.
2189 It is NOT used by default, as duplicating the behaviour of the old pattern
2190 matcher is still under development. A config file option, in extensions.conf,
2191 in the [general] section, called "extenpatternmatchingnew", is by default
2192 set to false; setting that to true will force the use of the new algorithm.
2193 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
2194 be used to switch the algorithms at run time.
2195 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
2196 specifying which socket to use to connect to the running Asterisk daemon
2198 * Performance enhancements to the sched facility, which is used in
2199 the channel drivers, etc. Added hashtabs and doubly-linked lists
2200 to speed up deletion; start at the beginning or end of list to
2202 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
2203 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
2204 Added regression tests to the tests/ dir, also.
2205 * Added a refcount trace feature to astobj2 for those trying to balance
2206 object creation, deletion; work, play; space and time. See the
2207 notes in astobj2.h. Also, see utils/refcounter as well, as a
2208 quick way to find unbalanced refcounts in what could be a sea
2209 of objects that were balanced.
2210 * Added logging to 'make update' command. See update.log
2211 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
2212 do not come from the remote party.
2213 * Added the 'n' option to the SpeechBackground application to tell it to not
2214 answer the channel if it has not already been answered.
2215 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
2216 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
2218 * iLBC source code no longer included (see UPGRADE.txt for details)
2219 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
2220 deadlock is detected, a backtrace of the stack which led to the lock calls
2221 will be output to the CLI.
2222 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
2223 the "core show locks" CLI command will give lock information output as well
2224 as a backtrace of the stack which led to the lock calls.
2225 * users.conf now sports an optional alternateexts property, which permits
2226 allocation of additional extensions which will reach the specified user.
2227 * A new option for the configure script, --enable-internal-poll, has been added
2228 for use with systems which may have a buggy implementation of the poll system
2229 call. If you notice odd behavior such as the CLI being unresponsive on remote
2230 consoles, you may want to try using this option. This option is enabled by default
2231 on Darwin systems since it is known that the Darwin poll() implementation has
2235 --------------------
2236 * In addition to timing from DAHDI, there is a new timing module called
2237 res_timing_timerfd. In order to use this, you must be running Linux with
2238 a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
2239 script will be able to tell if you have the requirements. From menuselect, select
2240 res_timing_timerfd from the Resource Modules menu.
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