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
49 * Named ACLs can now be specified in acl.conf and used in configurations that
50 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
51 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
52 working ACL. In addition, some CLI commands have
53 been added to provide informational and configuration reload capabilities to
54 this feature ('acl show [named acl]' and 'reload acl').
55 * Hangup handlers can be attached to channels using the CHANNEL(hangup_handler_xxx)
56 options. Hangup handlers will run when the channel is hung up similar to the
58 * The AMI Hangup event now includes the AccountCode header so you can easily
59 correlate with AMI Newchannel events.
63 * mixmonitor list <channel> command will now show MixMonitor ID, and the filenames
64 of all running mixmonitors on a channel.
65 * The debuglevel of "pri set debug" is now a bitmask ranging from 0 to 15 if
66 numeric instead of 0, 1, or 2.
67 * "stun show status" will show a table describing how the STUN client is behaving.
71 * Added menu action admin_toggle_mute_participants. This will mute / unmute
72 all non-admin participants on a conference. The confbridge configuration file
73 also allows for the default sounds played to all conference users when this
74 occurs to be overriden using sound_participants_unmuted and sound_participants_muted.
75 * Added menu action participant_count. This will playback the number of current
76 participants in a conference.
77 * Added announcement configuration option to user profile. If set the sound file will
78 be played to the user, and only the user, upon joining the conference bridge.
82 * Addition of the VM_INFO function - see Dialplan function changes
83 * The imapserver, imapport, and imapflags configuration options can now be
84 overriden on a user by user basis.
88 * Asterisk will no longer substitute CID number for CID name into display
89 name field if CID number exists without a CID name. This change improves
90 compatibility with certain device features such as Avaya IP500's directory
92 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
93 created using that setting to not be removed during SIP reload.
94 * Add support to realtime for the 'callbackextension' option
95 * When multiple peers exist with the same address, but differing
96 callbackextension options, incoming requests that are matched by address
97 will be matched to the peer with the matching callbackextension if it is
99 * NAT settings are now a combinable list of options. The equivalent of the
100 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
101 * Two new NAT options, auto_force_rport and auto_comedia, have been added
102 which set the force_rport and comedia options automatically if Asterisk
103 detects that an incoming SIP request crossed a NAT after being sent by
105 * Adds an option send_diversion which can be disabled to prevent
106 diversion headers from automatically being added to invites.
107 * Add support for lightweight NAT keepalive. If enabled a blank packet will
108 be sent to the remote host at a given interval to keep the NAT mapping open.
109 This can be enabled using the keepalive configuration option.
110 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
115 * Added a manager event "LocalBridge" for local channel call bridges between
116 the two pseudo-channels created.
120 * Added dialtone_detect option for analog ports to disconnect incoming
121 calls when dialtone is detected.
123 ------------------------------------------------------------------------------
124 --- Functionality changes since Asterisk 10.4.0 ------------------------------
125 ------------------------------------------------------------------------------
129 * The optimization portion of the build system has been reworked to avoid
130 broken builds on certain architectures. All architecture-specific
131 optimization has been removed in favor of using -march=native to allow gcc
132 to detect the environment in which it is running when possible. This can
133 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
135 ------------------------------------------------------------------------------
136 --- Functionality changes since Asterisk 10.3.0 ------------------------------
137 ------------------------------------------------------------------------------
141 * Added ability to use multiple lines on phone, so for one device in
142 configuration multiple lines can be defined, it allows to have multiple calls
143 on one phone, callwaiting and switching between calls.
144 * Added option 'sharpdial' allowing end dialing by pressing # key
145 * Added option 'interdigit_timer' for controll phone dial timeout
146 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
147 * Added global 'debug' option, that enables debug in channel driver
148 * Added ability for translation on-screen menu to multiple languages. Tested on
149 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
150 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
152 * In addition to English added French and Russian languages for on-screen menus
153 * Reworked dialing number input: added dialing by timeout, immediate dial on
154 on dialplan compare, phone number length now not limited by screen size
155 * Added ability for pickup a call using features.conf defined value and
160 * Codec lists may now be modified by the '!' character, to allow succinct
161 specification of a list of codecs allowed and disallowed, without the
162 requirement to use two different keywords. For example, to specify all
163 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
165 Music On Hold Changes
166 ---------------------
167 * Added 'announcement' option which will play at the start of MOH and between
168 songs in modes of MOH that can detect transitions between songs (eg.
173 * Added queue options autopausebusy and autopauseunavail for automatically
174 pausing a queue member when their device reports busy or congestion.
175 * The 'ignorebusy' option for queue members has been deprecated in favor of
176 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
177 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
178 per interface basis. Individual ringinuse values can now be set in
179 queues.conf via an argument to member definitions. Lastly, the queue
180 'ringinuse' setting now only determines defaults for the per member
181 'ringinuse' setting and does not override per member settings like it does
186 * When voicemail plays a message's envelope with saycid set to yes, when reaching
187 the caller id field it will play a recording of a file with the same base name
188 as the sender's callerid if there is a similarly named file in
189 <astspooldir>/recordings/callerids/
193 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
194 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
195 changed arguments to SayUnixTime so that every option is truly optional even
196 when using multiple options (so that j option could be used without having to
197 manually specify timezone and format) There are other beneftis eg. format can
198 now be used without specifying time zone as well.
199 * Added 'F()' option to Queue and Bridge. Similar to the dial option, these can
200 be supplied with arguments indicating where the callee should go after the caller
201 is hung up, or without options specified, the priority after the Queue/Bridge
203 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
204 channels respectively before the callee channels are called.
208 * New per parking lot options: comebackcontext and comebackdialtime. See
209 configs/features.conf.sample for more details.
211 * Channel variable PARKER is now set when comebacktoorigin is disabled in
214 * MixMonitor hooks now have IDs associated with them which can be used to assign
215 a target to StopMixMonitor. Use of MixMonitor's i(variable) option will allow
216 storage of the MixMontior ID in a channel variable. StopMixmonitor now accepts
217 that ID as an argument.
219 CDR postgresql driver changes
220 -----------------------------
221 * Added command "cdr show pgsql status" to check connection status
223 AMI (Asterisk Manager Interface) changes
224 ----------------------------------------
225 * Originate now generates an error response if the extension given
226 is not found in the dialplan
228 * MixMonitor will now show IDs associated with the mixmonitor upon creating them
229 if the i(variable) option is used. StopMixMonitor will accept MixMonitorID as
230 on option to close specific MixMonitors.
232 * The SIPshowpeer manager action response field "SIP-Forcerport" has been updated
233 to include information about peers configured with nat=auto_force_rport by
234 returning "A" if auto_force_rport is set and nat is detected, and "a" if it is
235 set and nat is not detected. "Y" and "N" are still returned if auto_force_rport
238 * Hangup now can take a regular expression as the Channel option. If you want
239 to hangup multiple channels, use /regex/ as the Channel option. Existing
240 behavior to hanging up a single channel is unchanged, but if you pass a regex,
241 the manager will send you a list of channels back that were hung up.
243 * Support for IPv6 addresses has been added.
245 * AMI Events can now be documented in the Asterisk source. Two new CLI
246 commands have been added to display information about AMI events at run time:
247 manager show events, which shows a list of all known and documented AMI
248 events, and manager show event [event name], which shows detail information
249 about a specific AMI event. Note that AMI event documentation is only
250 generated when Asterisk is compiled using 'make full'.
254 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
255 control of faxdetect.
259 * Allow the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to be
260 used within the dynamic weight attribute when specifying a mapping.
264 * Addition of the VM_INFO function that can be used to retrieve voicemail
265 user information, such as the email address and full name.
266 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
268 * The REDIRECTING function now supports the redirecting original party id
270 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
271 lets you set some of the configuration options from the [general] section
272 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
273 the key sequence used to activate built-in features, such as blindxfer,
274 and automon. See the built-in documentation for details.
278 * A new option, 'I' has been added to app_followme.
279 By setting this option, Asterisk will not update the caller with
280 connected line changes when they occur. This is similar to app_dial
282 * The 'N' option is now ignored if the call is already answered.
283 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
284 and caller channels respectively before the callee channels are called.
288 * A new option, 'probation' has been added to rtp.conf
289 RTP in strictrtp mode can now require more than 1 packet to exit learning
290 mode with a new source (and by default requires 4). The probation option
291 allows the user to change the required number of packets in sequence to any
292 desired value. Use a value of 1 to essentially restore the old behavior.
293 Also, with strictrtp on, Asterisk will now drop all packets until learning
294 mode has successfully exited. These changes are based on how pjmedia handles
295 media sources and source changes.
299 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
300 instead of simply the uri. This is the format that MessageSend() can use
301 in the from parameter for outgoing SIP messages.
305 * A new module, res_corosync, has been introduced. This module uses the
306 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
307 of Asterisk servers to both Message Waiting Indication (MWI) and/or
308 Device State (presence) information. This module is very similar to, and
309 is a replacement for the res_ais module that was in previous releases of
314 * A new channel variable, AGIEXITONHANGUP, has been added which allows
315 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
316 AGI application would exit immediately after a channel hangup is detected.
317 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
318 are resolved and each address is attempted in turn until one succeeds or
323 * Direct media functionality has been added.
324 Options in config are: directmedia (directrtp) and directrtpsetup (earlydirect)
328 * A new channel driver named chan_motif has been added which provides support for
329 Google Talk and Jingle in a single channel driver. This new channel driver includes
330 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
331 hold, unhold, and ringing notification. It is also compliant with the current Jingle
332 specification, current Google Jingle specification, and the original Google Talk
335 ------------------------------------------------------------------------------
336 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
337 ------------------------------------------------------------------------------
341 * Asterisk now has protocol independent support for processing text messages
342 outside of a call. Messages are routed through the Asterisk dialplan.
343 SIP MESSAGE and XMPP are currently supported. There are options in
344 jabber.conf and sip.conf to allow enabling these features.
345 -> jabber.conf: see the "sendtodialplan" and "context" options.
346 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
347 and "outofcall_message_context" options.
348 The MESSAGE() dialplan function and MessageSend() application have been
349 added to go along with this functionality. More detailed usage information
350 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
351 * If real-time text support (T.140) is negotiated, it will be preferred for
352 sending text via the SendText application. For example, via SIP, messages
353 that were once sent via the SIP MESSAGE request would be sent via RTP if
354 T.140 text is negotiated for a call.
358 * parkedmusicclass can now be set for non-default parking lots.
360 Asterisk Manager Interface
361 --------------------------
362 * PeerStatus now includes Address and Port.
363 * Added Hold events for when the remote party puts the call on and off hold
364 for chan_dahdi ISDN channels.
365 * Added new action MeetmeListRooms to list active conferences (shows same
366 data as "meetme list" at the CLI).
367 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
368 Description field that is set by 'description' in the channel configuration
370 * Added Uniqueid header to UserEvent.
371 * Added new action FilterAdd to control event filters for the current session.
372 This requires the system permission and uses the same filter syntax as
373 filters that can be defined in manager.conf
374 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
375 versions had some instances of the event converted, but others were left
376 as-is. All Unlink events should now be converted to Bridge events. The AMI
377 protocol version number was incremented to 1.2 as a result of this change.
380 --------------------------
381 * The HTTP Server can bind to IPv6 addresses.
384 --------------------------
385 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
386 with busydetect. usage example: busypattern=200,200,200,600
389 --------------------------
390 * New 'gtalk show settings' command showing the current settings loaded from
392 * The 'logger reload' command now supports an optional argument, specifying an
393 alternate configuration file to use.
394 * 'dialplan add extension' command will now automatically create a context if
395 the specified context does not exist with a message indicated it did so.
396 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
397 Description field which can be populated with 'description' in the channel
398 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
401 --------------------------
402 * The filter option in cdr_adaptive_odbc now supports negating the argument,
403 thus allowing records which do NOT match the specified filter.
404 * Added ability to log CONGESTION calls to CDR
407 --------------------------
408 * Ability to define custom SILK formats in codecs.conf.
409 * Addition of speex32 audio format with translation.
410 * CELT codec pass-through support and ability to define
411 custom CELT formats in codecs.conf.
412 * Ability to read raw signed linear files with sample rates
413 ranging from 8khz - 192khz. The new file extensions introduced
414 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
415 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
416 Skinny, H.323, etc) can still only support the following codecs:
417 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
418 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
419 Video: h261, h263, h263p, h264, mpeg4
424 --------------------------
425 * New highly optimized and customizable ConfBridge application capable of
426 mixing audio at sample rates ranging from 8khz-96khz.
427 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
428 and bridge profiles on a channel.
429 * CONFBRIDGE_INFO dialplan function capable of retrieving information
430 about a conference such as locked status and number of parties, admins,
432 * Addition of video_mode option in confbridge.conf for adding video support
433 into a bridge profile.
434 * Addition of the follow_talker video_mode in confbridge.conf. This video
435 mode dynamically switches the video feed to always display the loudest talker
436 supplying video in the conference.
440 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
441 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
442 variables from asterisk.conf.
446 * Addition of the JITTERBUFFER dialplan function. This function allows
447 for jitterbuffering to occur on the read side of a channel. By using
448 this function conference applications such as ConfBridge and MeetMe can
449 have the rx streams jitterbuffered before conference mixing occurs.
450 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
452 * Added STRREPLACE function. This function let's the user search a variable
453 for a given string to replace with another string as many times as the
454 user specifies or just throughout the whole string.
455 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
456 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
457 * Added extensions to chan_ooh323 in function CHANNEL()
459 libpri channel driver (chan_dahdi) DAHDI changes
460 --------------------------
461 * Added moh_signaling option to specify what to do when the channel's bridged
462 peer puts the ISDN channel on hold.
463 * Added display_send and display_receive options to control how the display ie
464 is handled. To send display text from the dialplan use the SendText()
465 application when the option is enabled.
466 * Added mcid_send option to allow sending a MCID request on a span.
469 --------------------------
470 * Added setvar option to calendar.conf to allow setting channel variables on
471 notification channels.
472 * Added "calendar show types" CLI command to list registered calendar
476 --------------------------
477 * Added two new options, r and t with file name arguments to record
478 single direction (unmixed) audio recording separate from the bidirectional
479 (mixed) recording. The mixed file name argument is optional now as long
480 as at least one recording option is used.
483 --------------------------
484 * Added a new option, l, which will disable local call optimization for
485 channels involved with the FollowMe thread. Use this option to improve
486 compatability for a FollowMe call with certain dialplan apps, options, and
490 --------------------------
491 * Added option "k" that will automatically close the conference when there's
492 only one person left when a user exits the conference.
495 --------------------------
496 * cel_pgsql now supports the 'extra' column for data added using the
497 CELGenUserEvent() application.
500 --------------------------
501 * Support for defining hints has been added to pbx_lua. See the 'hints' table
502 in the sample extensions.lua file for syntax details.
503 * Applications that perform jumps in the dialplan such as Goto will now
504 execute properly. When pbx_lua detects that the context, extension, or
505 priority we are executing on has changed it will immediately return control
506 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
507 the priority after the currently executing priority.
508 * An autoservice is now started by default for pbx_lua channels. It can be
509 stopped and restarted using the autoservice_stop() and autoservice_start()
513 --------------------------
514 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
515 into a FAXStatus event with an 'Operation' header that will be either
516 'send', 'receive', and 'gateway'.
517 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
518 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
519 feature will handle converting a fax call between an audio T.30 fax terminal
520 and an IFP T.38 fax terminal.
524 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
525 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
526 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
530 * Added general option negative_penalty_invalid default off. when set
531 members are seen as invalid/logged out when there penalty is negative.
532 for realtime members when set remove from queue will set penalty to -1.
533 * Added queue option autopausedelay when autopause is enabled it will be
534 delayed for this number of seconds since last successful call if there
535 was no prior call the agent will be autopaused immediately.
536 * Added member option ignorebusy this when set and ringinuse is not
537 will allow per member control of multiple calls as ringinuse does for
539 * Added global option check_state_unknown to enforce checking of device state
540 when the device state is unknown app_queue will see unknown as available.
544 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
546 * Added 'k' option to MeetMe to automatically kill the conference when there's only
547 one participant left (much like a normal call bridge)
548 * Added extra argument to Originate to set timeout.
552 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
553 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
554 utility in the UTILS section of menuselect. If an existing astdb is found and no
555 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
556 convert an existing astdb to the SQLite3 version automatically at runtime.
560 * Modules marked as deprecated are no longer marked as building by default. Enabling
561 these modules is still available via menuselect.
565 * authdebug is now disabled by default. To enable this functionaility again
566 set authdebug = yes in iax.conf.
570 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
571 releases it was disabled.
575 * The PBX core previously made a call with a non-existing extension test for
576 extension s@default and jump there if the extension existed.
577 This was a bad default behaviour and violated the principle of least surprise.
578 It has therefore been changed in this release. It may affect some
579 applications and configurations that rely on this behaviour. Most channel
580 drivers have avoided this for many releases by testing whether the extension
581 called exists before starting the PBX and generating a local error.
582 This behaviour still exists and works as before.
584 Extension "s" is used when no extension is given in a channel driver,
585 like immediate answer in DAHDI or calling to a domain with no user part
588 ------------------------------------------------------------------------------
589 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
590 ------------------------------------------------------------------------------
594 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
595 now defaults to force_rport. It is very important that phones requiring nat=no be
596 specifically set as such instead of relying on the default setting. If at all
597 possible, all devices should have nat settings configured in the general section as
598 opposed to configuring nat per-device.
599 * Added preferred_codec_only option in sip.conf. This feature limits the joint
600 codecs sent in response to an INVITE to the single most preferred codec.
601 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
602 to be used for the outgoing call. It must be one of the codecs configured
604 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
605 to be used for holding a private key. If tlsprivatekey is not specified,
606 tlscertfile is searched for both public and private key.
607 * Added tlsclientmethod option to sip.conf. This allows the protocol for
608 outbound client connections to be specified.
609 * The sendrpid parameter has been expanded to include the options
610 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
611 header to be sent (equivalent to setting sendrpid=yes) and setting
612 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
613 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
614 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
615 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
616 will accept the SDP even if the SDP version number is not properly incremented,
617 but will generate a warning in the log indicating that the SIP peer that sent
618 the SDP should have the 'ignoresdpversion' option set.
619 * The 'nat' option has now been been changed to have yes, no, force_rport, and
620 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
621 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
622 remote side requests it and disables symmetric RTP support. Setting it to
623 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
624 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
625 and enables symmetric RTP support.
626 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
627 response. This permits the master channel to know how each channel dialled
628 in a multi-channel setup resolved in an individual way. This carries a
629 performance penalty and can be disabled in sip.conf using the
630 'storesipcause' option.
631 * Added 'externtcpport' and 'externtlsport' options to allow custom port
632 configuration for the externip and externhost options when tcp or tls is used.
633 * Added support for message body (stored in content variable) to SIP NOTIFY message
634 accessible via AMI and CLI.
635 * Added 'media_address' configuration option which can be used to explicitly specify
636 the IP address to use in the SDP for media (audio, video, and text) streams.
637 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
638 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
640 * Added 'use_q850_reason' configuration option for generating and parsing
641 if available Reason: Q.850;cause=<cause code> header. It is implemented
642 in some gateways for better passing PRI/SS7 cause codes via SIP.
643 * When dialing SIP peers, a new component may be added to the end of the dialstring
644 to indicate that a specific remote IP address or host should be used when dialing
645 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
646 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
647 ability to selectively force bridged channels to also be encrypted is also
648 implemented. Branching in the dialplan can be done based on whether or not
649 a channel has secure media and/or signaling.
650 * Added directmediapermit/directmediadeny to limit which peers can send direct media
652 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
653 Charge messages to snom phones.
654 * Added support for G.719 media streams.
655 * Added support for 16khz signed linear media streams.
656 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
657 RTP has been outfitted with the same abilities.
658 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
659 available in device configurations as well as in the dial plan.
660 * Addition of the 'subscribe_network_change' option for turning on and off
661 res_stun_monitor module support in chan_sip.
662 * Addition of the 'auth_options_requests' option for turning on and off
663 authentication for OPTIONS requests in chan_sip.
667 * Add #tryinclude statement for config files. This provides the same
668 functionality as the #include statement however an asterisk module will
669 still load if the filename does not exist. Using the #include statement
670 Asterisk will not allow the module to load.
674 * Added rtsavesysname option into iax.conf to allow the systname to be saved
676 * Added the ability for chan_iax2 to inform the dialplan whether or not
677 encryption is being used. This interoperates with the SIP SRTP implementation
678 so that a secure SIP call can be bridged to a secure IAX call when the
679 dialplan requires bridged channels to be "secure".
680 * Addition of the 'subscribe_network_change' option for turning on and off
681 res_stun_monitor module support in chan_iax.
686 * Added ability to preset channel variables on indicated lines with the setvar
687 configuration option. Also, clearvars=all resets the list of variables back
689 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
690 See configs/res_pktccops.conf for more information.
692 XMPP Google Talk/Jingle changes
693 -------------------------------
694 * Added the externip option to gtalk.conf.
695 * Added the stunaddr option to gtalk.conf which allows for the automatic
696 retrieval of the external ip from a stun server.
700 * Added 'p' option to PickupChan() to allow for picking up channel by the first
701 match to a partial channel name.
702 * Added .m3u support for Mp3Player application.
703 * Added progress option to the app_dial D() option. When progress DTMF is
704 present, those values are sent immediately upon receiving a PROGRESS message
705 regardless if the call has been answered or not.
706 * Added functionality to the app_dial F() option to continue with execution
707 at the current location when no parameters are provided.
708 * Added the 'a' option to app_dial to answer the calling channel before any
709 announcements or macros are executed.
710 * Modified app_dial to set answertime when the called channel answers even if
711 the called channel hangs up during playback of an announcement.
712 * Modified app_dial 'r' option to support an additional parameter to play an
713 indication tone from indications.conf
714 * Added c() option to app_chanspy. This option allows custom DTMF to be set
715 to cycle through the next available channel. By default this is still '*'.
716 * Added x() option to app_chanspy. This option allows DTMF to be set to
717 exit the application.
718 * The Voicemail application has been improved to automatically ignore messages
719 that only contain silence.
720 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
721 associated mailbox(es) to be greetings-only.
722 * The ChanSpy application now has the 'S' option, which makes the application
723 automatically exit once it hits a point where no more channels are available
725 * The ChanSpy application also now has the 'E' option, which spies on a single
726 channel and exits when that channel hangs up.
727 * The MeetMe application now turns on the DENOISE() function by default, for
728 each participant. In our tests, this has significantly decreased background
729 noise (especially noisy data centers).
730 * Voicemail now permits storage of secrets in a separate file, located in the
731 spool directory of each individual user. The control for this is located in
732 the "passwordlocation" option in voicemail.conf. Please see the sample
733 configuration for more information.
734 * The ChanIsAvail application now exposes the returned cause code using a separate
735 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
736 * Added 'd' option to app_followme. This option disables the "Please hold"
738 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
739 received will terminate recording.
740 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
741 Previously the folder could only be set per context, but has now been extended
742 using the imapfolder option.
743 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
744 * Voicemail now allows the pager date format to be specified separately from the
746 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
747 to allow joining, leaving, and sending text to group chats.
748 * MeetMe has a new option 'G' to play an announcement before joining a conference.
749 * Page has a new option 'A(x)' which will playback an announcement simultaneously
750 to all paged phones (and optionally excluding the caller's one using the new
751 option 'n') before the call is bridged.
752 * The 'f' option to Dial has been augmented to take an optional argument. If no
753 argument is provided, the 'f' option works as it always has. If an argument is
754 provided, then the connected party information of all outgoing channels created
755 during the Dial will be set to the argument passed to the 'f' option.
756 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
758 * The OSP lookup application adds in/outbound network ID, optional security,
759 number portability, QoS reporting, destination IP port, custom info and service
761 * Added new application VMSayName that will play the recorded name of the voicemail
762 user if it exists, otherwise will play the mailbox number.
763 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
764 retrieve state for a particular bridge, where <name> is the conference name
765 * app_directory now allows exiting at any time using the operator or pound key.
766 * Voicemail now supports setting a locale per-mailbox.
767 * Two new applications are provided for declining counting phrases in multiple
768 languages. See the application notes for SayCountedNoun and SayCountedAdj for
770 * Voicemail now runs the externnotify script when pollmailboxes is activated and
772 * Voicemail now includes rdnis within msgXXXX.txt file.
773 * ExternalIVR now supports IPv6 addresses.
774 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
775 at https://wiki.asterisk.org/wiki/x/oQBB
776 * ParkedCall and Park can now specify the parking lot to use.
780 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
781 over SRV records associated with a specific service. From the CLI, type
782 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
783 details on how these may be used.
784 * PITCH_SHIFT dialplan function added. This function can be used to modify the
785 pitch of a channel's tx and rx audio streams.
786 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
787 setting various connected line and redirecting party information.
788 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
789 support ISDN subaddressing.
790 * The CHANNEL() function now supports the "name" and "checkhangup" options.
791 * For DAHDI channels, the CHANNEL() dialplan function now allows
792 the dialplan to request changes in the configuration of the active
793 echo canceller on the channel (if any), for the current call only.
796 exten => s,n,Set(CHANNEL(echocan_mode)=off)
798 The possible values are:
800 on - normal mode (the echo canceller is actually reinitialized)
802 fax - FAX/data mode (NLP disabled if possible, otherwise completely
804 voice - voice mode (returns from FAX mode, reverting the changes that
805 were made when FAX mode was requested)
806 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
807 and setting variables on the channel which created the current channel.
808 Administrators should take care to avoid naming conflicts, when multiple
809 channels are dialled at once, especially when used with the Local channel
810 construct (which all could set variables on the master channel). Usage
811 of the HASH() dialplan function, with the key set to the name of the slave
812 channel, is one approach that will avoid conflicts.
813 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
815 * func_odbc now allows multiple row results to be retrieved without using
816 mode=multirow. If rowlimit is set, then additional rows may be retrieved
817 from the same query by using the name of the function which retrieved the
818 first row as an argument to ODBC_FETCH().
819 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
820 dialplan. This function returns the content of the received message.
821 * Added REPLACE, which searches a given variable name for a set of characters,
822 then either replaces them with a single character or deletes them.
823 * Added PASSTHRU, which literally passes the same argument back as its return
824 value. The intent is to be able to use a literal string argument to
825 functions that currently require a variable name as an argument.
826 * HASH-associated variables now can be inherited across channel creation, by
827 prefixing the name of the hash at assignment with the appropriate number of
828 underscores, just like variables.
829 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
830 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
831 whether or not channels that are bridged to the current channel will be
832 required to have secure signaling and/or media.
833 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
834 the current channel has secure signaling and/or media.
835 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
836 "no_media_path" option.
837 Returns "0" if there is a B channel associated with the call.
838 Returns "1" if no B channel is associated with the call. The call is either
839 on hold or is a call waiting call.
840 * Added option to dialplan function CDR(), the 'f' option
841 allows for high resolution times for billsec and duration fields.
842 * FILE() now supports line-mode and writing.
843 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
844 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
848 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
849 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
850 and is set when a dynamic feature is triggered.
851 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
852 to dynamically create a new parking lot matching the value this varible is
854 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
855 features.conf that should be the base for dynamic parkinglots.
856 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
857 parkinglot should have.
858 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
859 parkinglot should have.
860 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
865 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
867 * Added 'R' option to app_queue. This option stops moh and indicates ringing
868 to the caller when an Agent's phone is ringing. This can be used to indicate
869 to the caller that their call is about to be picked up, which is nice when
870 one has been on hold for an extened period of time.
871 * A new config option, penaltymemberslimit, has been added to queues.conf.
872 When set this option will disregard penalty settings when a queue has too
874 * A new option, 'I' has been added to both app_queue and app_dial.
875 By setting this option, Asterisk will not update the caller with
876 connected line changes or redirecting party changes when they occur.
877 * A 'relative-periodic-announce' option has been added to queues.conf. When
878 enabled, this option will cause periodic announce times to be calculated
879 from the end of announcements rather than from the beginning.
880 * The autopause option in queues.conf can be passed a new value, "all." The
881 result is that if a member becomes auto-paused, he will be paused in all
882 queues for which he is a member, not just the queue that failed to reach
884 * Added dialplan function QUEUE_EXISTS to check if a queue exists
885 * The queue logger now allows events to optionally propagate to a file,
886 even when realtime logging is turned on. Additionally, realtime logging
887 supports sending the event arguments to 5 individual fields, although it
888 will fallback to the previous data definition, if the new table layout is
891 mISDN channel driver (chan_misdn) changes
892 ----------------------------------------
893 * Added display_connected parameter to misdn.conf to put a display string
894 in the CONNECT message containing the connected name and/or number if
895 the presentation setting permits it.
896 * Added display_setup parameter to misdn.conf to put a display string
897 in the SETUP message containing the caller name and/or number if the
898 presentation setting permits it.
899 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
900 indicate the dialplan settings are to be obtained from the asterisk
902 * Made misdn.conf parameter callerid accept the "name" <number> format
903 used by the rest of the system.
904 * Made use the nationalprefix and internationalprefix misdn.conf
905 parameters to prefix any received number from the ISDN link if that
906 number has the corresponding Type-Of-Number. NOTE: This includes
907 comparing the incoming call's dialed number against the MSN list.
908 * Added the following new parameters: unknownprefix, netspecificprefix,
909 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
910 received number from the ISDN link if that number has the corresponding
912 * Added new dialplan application misdn_command which permits controlling
913 the CCBS/CCNR functionality.
914 * Added new dialplan function mISDN_CC which permits retrieval of various
915 values from an active call completion record.
916 * For PTP, you should manually send the COLR of the redirected-to party
917 for an incomming redirected call if the incoming call could experience
918 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
919 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
920 if the REDIRECTING(from-num) is not empty.
921 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
922 option on all of the REDIRECTING statements before dialing the
923 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
924 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
925 redirecting-to presentation (COLR) when it becomes available.
926 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
929 thirdparty mISDN enhancements
930 -----------------------------
931 mISDN has been modified by Digium, Inc. to greatly expand facility message
933 * Enhanced COLP support for call diversion and transfer.
936 The latest modified mISDN v1.1.x based version is available at:
937 http://svn.digium.com/svn/thirdparty/mISDN/trunk
938 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
940 Tagged versions of the modified mISDN code are available under:
941 http://svn.digium.com/svn/thirdparty/mISDN/tags
942 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
944 libpri channel driver (chan_dahdi) DAHDI changes
945 -------------------------------------------
946 * The channel variable PRIREDIRECTREASON is now just a status variable
947 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
948 to read and alter the reason.
949 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
950 redirected-to party for an incomming redirected call if the incoming call
951 could experience further redirects. Just set the
952 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
953 to the COLR. A call has been redirected if the REDIRECTING(count) is not
955 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
956 use the inhibit(i) option on all of the REDIRECTING statements before
957 dialing the redirected-to party. You still have to set the
958 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
959 will update the redirecting-to presentation (COLR) when it becomes available.
960 * Added the ability to ignore calls that are not in a Multiple Subscriber
961 Number (MSN) list for PTMP CPE interfaces.
962 * Added dynamic range compression support for dahdi channels. It is
963 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
964 * Added support for ISDN calling and called subaddress with partial support
965 for connected line subaddress.
966 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
967 * Added handling of received HOLD/RETRIEVE messages and the optional ability
968 to transfer a held call on disconnect similar to an analog phone.
969 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
970 Will reroute/deflect an outgoing call when receive the message.
971 Can use the DAHDISendCallreroutingFacility to send the message for the
973 * Added standard location to add options to chan_dahdi dialing:
974 Dial(DAHDI/g1[/extension[/options]])
977 R Reverse charging indication
978 * Added Reverse Charging Indication (Collect calls) send/receive option.
979 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
980 Dial(DAHDI/g1/extension/R)
981 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
982 (requires latest LibPRI)
983 * Added ability to send/receive keypad digits in the SETUP message.
984 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
985 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
986 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
987 (requires latest LibPRI)
988 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
989 to eliminate tromboned calls. A tromboned call goes out an interface and comes
990 back into the same interface. Tromboned calls happen because of call routing,
991 call deflection, call forwarding, and call transfer.
992 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
993 * Added the ability to support call waiting calls. (The SETUP has no B channel
995 * Added Malicious Call ID (MCID) event to the AMI call event class.
996 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
998 Asterisk Manager Interface
999 --------------------------
1000 * The Hangup action now accepts a Cause header which may be used to
1001 set the channel's hangup cause.
1002 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
1003 to specify a separate .pem file to hold a private key. By default sslcert
1004 is used to hold both the public and private key.
1005 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
1006 for options containing the 'tls' prefix. For example, 'sslenable' is now
1007 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
1008 across all .conf files. All affected sample.conf files have been modified to
1009 reflect this change. Previous options such as 'sslenable' still work,
1010 but options with the 'tls' prefix are preferred.
1011 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
1012 in a channel. (res_mutestream.so)
1013 * The configuration file manager.conf now supports a channelvars option, which
1014 specifies a list of channel variables to include in each channel-oriented
1016 * The redirect command now has new parameters ExtraContext, ExtraExtension,
1017 and ExtraPriority to allow redirecting the second channel to a different
1018 location than the first.
1019 * Added new event "JabberStatus" in the Jabber module to monitor buddies
1021 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
1022 in a MixMonitor recording.
1023 * The 'iax2 show peers' output is now similar to the expected output of
1025 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
1027 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
1028 AOC-E messages on a channel.
1029 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
1030 conform more closely to similar events.
1031 * Added a new eventfilter option per user to allow whitelisting and blacklisting
1033 * Added optional parkinglot variable for park command.
1034 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
1035 if CallerIDNum and CallerIDName headers are also present.
1037 Channel Event Logging
1038 ---------------------
1039 * A new interface, CEL, is introduced here. CEL logs single events, much like
1040 the AMI, but it differs from the AMI in that it logs to db backends much
1041 like CDR does; is based on the event subsystem introduced by Russell, and
1042 can share in all its benefits; allows multiple backends to operate like CDR;
1043 is specialized to event data that would be of concern to billing sytems,
1044 like CDR. Backends for logging and accounting calls have been produced,
1045 but a new CDR backend is still in development.
1049 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
1050 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
1051 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
1052 * Multiple files and formats can now be specified in cdr_custom.conf.
1053 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
1054 See configs/cdr_syslog.conf.sample for more information.
1055 * A 'sequence' field has been added to CDRs which can be combined with
1056 linkedid or uniqueid to uniquely identify a CDR.
1057 * Handling of billsec and duration field has changed. If your table definition
1058 specifies those fields as float,double or similar they will now be logged with
1059 microsecond accuracy instead of a whole integer.
1061 Calendaring for Asterisk
1062 ------------------------
1063 * A new set of modules were added supporing calendar integration with Asterisk.
1064 Dialplan functions for reading from and writing to calendars are included,
1065 as well as the ability to execute dialplan logic upon calendar event notifications.
1066 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
1067 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
1068 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
1069 2003 support does not support forms-based authentication).
1071 Call Completion Supplementary Services for Asterisk
1072 ---------------------------------------------------
1073 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
1074 DAHDI/ISDN supports call completion for the following switch types:
1075 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
1076 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
1078 Multicast RTP Support
1079 ---------------------
1080 * A new RTP engine and channel driver have been added which supports Multicast RTP.
1081 The channel driver can be used with the Page application to perform multicast RTP
1082 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
1083 Type can be either basic or linksys.
1084 Destination is the IP address and port for the RTP packets.
1085 Control address is specific to the linksys type and is used for sending the control
1086 packets unique to them.
1088 Security Events Framework
1089 -------------------------
1090 * Asterisk has a new C API for reporting security events. The module res_security_log
1091 sends these events to the "security" logger level. Currently, AMI is the only
1092 Asterisk component that reports security events. However, SIP support will be
1093 coming soon. For more information on the security events framework, see the
1094 "Asterisk Security Framework" section of the Asterisk wiki at
1095 https://wiki.asterisk.org/wiki/x/wgBQ
1096 * SIP support was added in Asterisk 10
1097 * This API now supports IPv6 addresses
1101 * A technology independent fax frontend (res_fax) has been added to Asterisk.
1102 * A spandsp based fax backend (res_fax_spandsp) has been added.
1103 * The app_fax module has been deprecated in favor of the res_fax module and
1104 the new res_fax_spandsp backend.
1105 * The SendFAX and ReceiveFAX applications now send their log messages to a
1106 'fax' logger level, instead of to the generic logger levels. To see these
1107 messages, the system's logger.conf file will need to direct the 'fax' logger
1108 level to one or more destinations; the logger.conf.sample file includes an
1109 example of how to do this. Note that if the 'fax' logger level is *not*
1110 directed to at least one destination, log messages generated by these
1111 applications will be lost, and that if the 'fax' logger level is directed to
1112 the console, the 'core set verbose' and 'core set debug' CLI commands will
1113 have no effect on whether the messages appear on the console or not.
1117 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
1118 Now, in order to enable transmitting silence during record the transmit_silence
1119 option should be used. transmit_silence_during_record remains a valid option, but
1120 defaults to the behavior of the transmit_silence option.
1121 * Addition of the Unit Test Framework API for managing registration and execution
1122 of unit tests with the purpose of verifying the operation of C functions.
1123 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
1124 XMPP text messages to the remote JID.
1125 * Modules.conf has a new option - "require" - that marks a module as critical for
1126 the execution of Asterisk.
1127 If one of the required modules fail to load, Asterisk will exit with a return
1129 * An 'X' option has been added to the asterisk application which enables #exec support.
1130 This allows #exec to be used in asterisk.conf.
1131 * jabber.conf supports a new option auth_policy that toggles auto user registration.
1132 * A new lockconfdir option has been added to asterisk.conf to protect the
1133 configuration directory (/etc/asterisk by default) during reloads.
1134 * The parkeddynamic option has been added to features.conf to enable the creation
1135 of dynamic parkinglots.
1136 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
1137 the reportalarms config option.
1138 * chan_dahdi supports dialing configuring and dialing by device file name.
1139 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
1140 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
1141 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
1142 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
1143 Handy for the above name-based syntax as it does not depend on
1144 initialization order.
1145 * The Realtime dialplan switch now caches entries for 1 second. This provides a
1146 significant increase in performance (about 3X) for installations using this switchtype.
1147 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
1148 AIS. For more information, please see the Distributed Device State section of the
1149 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1150 * The addition of G.719 pass-through support.
1151 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
1152 during device configuration.
1153 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
1154 have less than 3 lines on the LCD.
1155 * Realtime now supports database failover. See the sample extconfig.conf for details.
1156 * The addition of improved translation path building for wideband codecs. Sample
1157 rate changes during translation are now avoided unless absolutely necessary.
1158 * The addition of the res_stun_monitor module for monitoring and reacting to network
1159 changes while behind a NAT.
1163 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
1164 optionally accept a filename, to apply the setting only to the code generated from
1165 that source file when Asterisk was built. However, there are some modules in Asterisk
1166 that are composed of multiple source files, so this did not result in the behavior
1167 that users expected. In this version, 'core set debug' and 'core set verbose'
1168 can optionally accept *module* names instead (with or without the .so extension),
1169 which applies the setting to the entire module specified, regardless of which source
1170 files it was built from.
1171 * New 'manager show settings' command showing the current settings loaded from
1173 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
1174 the channel hangup request to all channels.
1175 * Added a "core reload" CLI command that executes a global reload of Asterisk.
1177 ------------------------------------------------------------------------------
1178 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
1179 ------------------------------------------------------------------------------
1183 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
1184 Snom phones use this for call pickup of extensions that the phone is
1186 * Added support for setting the domain in the URI for caller of an
1187 outbound call by using the SIPFROMDOMAIN channel variable.
1188 * Added a new configuration option "remotesecret" for authentication to
1189 remote services. For backwards compatibility, "secret" still has the
1190 same function as before, but now you can configure both a remote secret and a
1191 local secret for mutual authentication.
1192 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
1193 the sound will be played to the target of an attended transfer
1194 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
1195 finer control over how many peers Asterisk will qualify and the gap between them
1196 when all peers need to be qualified at the same time.
1197 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
1198 (either globally or for a specific peer), chan_sip will treat any SDP data
1199 it receives as new data and update the media stream accordingly. By
1200 default, Asterisk will only modify the media stream if the SDP session
1201 version received is different from the current SDP session version. This
1202 option is required to interoperate with devices that have non-standard SDP
1203 session version implementations (observed with Microsoft OCS). This option
1204 is disabled by default.
1205 * The parsing of register => lines in sip.conf has been modified to allow a port
1206 to be present in the "user" portion. Please see the sip.conf.sample file for more
1208 * Added support for subscribing to MWI on a remote server and making the status available
1209 as a mailbox. Please see the sip.conf.sample file for more information.
1210 * Added a function to remove SIP headers added in the dialplan before the
1211 first INVITE is generated - SIPRemoveHeader()
1212 * Channel variables set with setvar= in a device configuration is now
1213 set both for inbound and outbound calls.
1214 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
1218 * Added immediate option to iax.conf
1219 * Added forceencryption option to iax.conf
1220 * Added Encryption and Trunk status to manager command "iaxpeers"
1224 * The configuration file now holds separate sections for devices and lines.
1225 Please have a look at configs/skinny.conf.sample and change your skinny.conf
1230 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
1231 support for LibOpenR2. http://www.libopenr2.org/
1232 * The UK option waitfordialtone has been added for use with BT analog
1234 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
1235 is used in conjunction with the 'faxdetect' configuration option. When
1236 'faxbuffers' is used and fax tones are detected, the channel will dynamically
1237 switch to the configured faxbuffers policy. For example, to use 6 buffers
1238 and a 'full' buffer policy for a fax transmission, add:
1240 The faxbuffers configuration will be in affect until the call is torn down.
1241 * Added service message support for 4ESS/5ESS switches.
1245 * For DAHDI channels, the CHANNEL() dialplan function now
1246 supports changing the channel's buffer policy (for the current
1247 call only), using this syntax:
1249 exten => s,n,Set(CHANNEL(buffers)=6,full)
1251 This would change the channel to the 'full' buffer policy and
1252 6 (six) buffers. Possible options for this setting are the same
1253 as those in chan_dahdi.conf.
1254 * Added a new dialplan function, CURLOPT, which permits setting various
1255 options that may be useful with the CURL dialplan function, such as
1256 cookies, proxies, connection timeouts, passwords, etc.
1257 * Permit the syntax and synopsis fields of the corresponding dialplan
1258 functions to be individually set from func_odbc.conf.
1259 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
1260 * func_odbc now may specify an insert query to execute, when the write query
1261 affects 0 rows (usually indicating that no such row exists).
1262 * Added a new dialplan function, LISTFILTER, which permits removing elements
1263 from a set list, by name. Uses the same general syntax as the existing CUT
1264 and FIELDQTY dialplan functions, which also manage lists.
1265 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
1266 obtaining realtime data from the dialplan.
1267 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
1268 a subroutine when using the GoSub() and Return() applications.
1269 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
1270 of "core show function AUDIOHOOK_INHERIT" from the CLI
1271 * Added AES_ENCRYPT. For information on its use, please see the output
1272 of "core show function AES_ENCRYPT" from the CLI
1273 * Added AES_DECRYPT. For information on its use, please see the output
1274 of "core show function AES_DECRYPT" from the CLI
1275 * func_odbc now supports database transactions across multiple queries.
1279 * Scheduled meetme conferences may now have their end times extended by
1281 * app_authenticate now gives the ability to select a prompt other than
1283 * app_directory now pays attention to the searchcontexts setting in
1284 voicemail.conf and will look through all contexts, if no context is
1285 specified in the initial argument.
1286 * A new application, Originate, has been introduced, that allows asynchronous
1287 call origination from the dialplan.
1288 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
1289 in addition to the setting in the "general" context.
1290 * Added ConfBridge dialplan application which does conference bridges without
1291 DAHDI. For information on its use, please see the output of
1292 "core show application ConfBridge" from the CLI.
1296 * The Asterisk CLI has a new command, "channel redirect", which is similar in
1297 operation to the AMI Redirect action.
1298 * extensions.conf now allows you to use keyword "same" to define an extension
1299 without actually specifying an extension. It uses exactly the same pattern
1300 as previously used on the last "exten" line. For example:
1301 exten => 123,1,NoOp(something)
1302 same => n,SomethingElse()
1303 * musiconhold.conf classes of type 'files' can now use relative directory paths,
1304 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
1305 * All deprecated CLI commands are removed from the sourcecode. They are now handled
1306 by the new clialiases module. See cli_aliases.conf.sample file.
1307 * Times within timespecs are now accurate down to the minute. This is a change
1308 from historical Asterisk, which only provided timespecs rounded to the nearest
1309 even (read: evenly divisible by 2) minute mark.
1310 * The realtime switch now supports an option flag, 'p', which disables searches for
1312 * In addition to a time range and date range, timespecs now accept a 5th optional
1313 argument, timezone. This allows you to perform time checks on alternate
1314 timezones, especially if those daylight savings time ranges vary from your
1315 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
1317 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
1318 give you the correct output for an asterisk box behind nat. It will give you the
1319 externhost and localnet settings.
1320 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
1321 can connect calls in passthrough mode, as well as record and play back files.
1322 * Successful and unsuccessful call pickup can now be alerted through sounds, by
1323 using pickupsound and pickupfailsound in features.conf.
1324 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
1325 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
1326 instead of the /var/run/asterisk.pid where it used to be. This will make
1327 installs as non-root easier to manage.
1332 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
1333 be written; they will no longer be explicitly written.
1335 Asterisk Manager Interface
1336 --------------------------
1337 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
1338 a non-empty value) in your request. If you do this, any pending AMI events will
1339 *not* be included in the response to your request as they would normally, but
1340 will be left in the event queue for the next request you make to retrieve. For
1341 some applications, this will allow you to guarantee that you will only see
1342 events in responses to 'WaitEvent' actions, and can better know when to expect them.
1343 To know whether the Asterisk server supports this header or not, your client can
1344 inspect the first response back from the server to see if it includes this header:
1346 Pragma: SuppressEvents
1348 If this is included, the server supports event suppression.
1350 * Added 4 new Actions to list skinny device(s) and line(s)
1356 LDAP Schema File Additions
1357 --------------------------
1358 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
1359 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
1361 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
1362 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
1363 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
1364 * Removed redundant IPaddr (there's already IPAddress)
1365 - Gives more configuration Flags for SIP-Users available (tested)
1366 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
1367 without extensibleObject (which really should be the last resort); gives
1368 also additional possibilities for LDAP-filter
1370 ------------------------------------------------------------------------------
1371 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
1372 ------------------------------------------------------------------------------
1374 Device State Handling
1375 ---------------------
1376 * The event infrastructure in Asterisk got another big update to help support
1377 distributed events. It currently supports distributed device state and
1378 distributed Voicemail MWI (Message Waiting Indication). A new module has
1379 been merged, res_ais, which facilitates communicating events between servers.
1380 It uses the SAForum AIS (Service Availability Forum Application Interface
1381 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
1382 a cluster of Asterisk servers, and to share events between them. For more
1383 information on setting this up, refer to the Distributed Device State section
1384 of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1388 * Added a new dialplan function, AST_CONFIG(), which allows you to access
1389 variables from an Asterisk configuration file.
1390 * The JACK_HOOK function now has a c() option to supply a custom client name.
1391 * Added two new dialplan functions from libspeex for audio gain control and
1392 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
1393 rx directions of a channel from the dialplan.
1394 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
1395 based on other parameters. The default is still to search based on the
1396 forwarding station ID. However, there are new options that allow you to search
1397 based on the message desk terminal ID, or the message desk number.
1398 * TIMEOUT() has been modified to be accurate down to the millisecond.
1399 * ENUM*() functions now include the following new options:
1400 - 'u' returns the full URI and does not strip off the URI-scheme.
1401 - 's' triggers ISN specific rewriting
1402 - 'i' looks for branches into an Infrastructure ENUM tree
1403 - 'd' for a direct DNS lookup without any flipping of digits.
1404 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
1405 * CHANNEL() now has options for the maximum, minimum, and standard or normal
1406 deviation of jitter, rtt, and loss for a call using chan_sip.
1408 DAHDI channel driver (chan_dahdi) Changes
1409 ----------------------------------------
1410 * Channels can now be configured using named sections in chan_dahdi.conf, just
1411 like other channel drivers, including the use of templates.
1412 * The default for pridialplan has changed from 'national' to 'unknown'.
1416 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
1417 to something that matches the pattern a hint will be created using the contents
1418 and variables evaluated.
1419 * Dialplan matching has been extended to allow an extension to return to the
1420 PBX core to wait for more digits. This is done by using the new dialplan
1421 application called "Incomplete". This will permit a whole new level of
1422 extension control, by giving the administrator more control over early
1423 matches employing one of the short-circuit pattern match operators. Note
1424 that custom applications can trigger this same behavior by returning the
1425 special value AST_PBX_INCOMPLETE.
1429 * Directory now permits both first and last names to be matched at the same
1430 time. In addition, the number of digits to enter of the name can be set in
1431 the arguments to Directory; previously, you could enter only 3, regardless
1432 of how many names are in your company. For large companies, this should be
1434 * Voicemail now permits a mailbox setting to wrap around from first to last
1435 messages, if the "messagewrap" option is set to a true value.
1436 * Voicemail now permits an external script to be run, for password validation.
1437 The script should output "VALID" or "INVALID" on stdout, depending upon the
1438 wish to validate or invalidate the password given. Arguments are:
1439 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
1441 * Dial has a new option: F(context^extension^pri), which permits a callee to
1442 continue in the dialplan, at the specified label, if the caller hangs up.
1443 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
1444 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
1445 * The Jack application now has a c() option to supply a custom client name.
1446 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
1447 like the pre-existing whisper mode, except that the spy can also talk to the
1448 participant on the bridged channel as well.
1449 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
1450 to be spoken instead of the channel name or number. For more information on the
1451 use of this option, issue the command "core show application ChanSpy" from the
1453 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
1454 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
1455 words, if using the 'd' option, it is not possible to enter a number to append to
1456 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
1457 change to whisper mode, and pressing 6 will change to barge mode.
1458 * ExternalIVR now takes several options that affect the way it performs, as
1459 well as having several new commands. Please see the External IVR page on the Asterisk
1460 wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
1461 * Added ability to communicate over a TCP socket instead of forking a child process for the
1462 ExternalIVR application.
1463 * ChanIsAvail has a new option, 'a', which will return all available channels instead
1464 of just the first one if you give the function more then one channel to check.
1465 * PrivacyManager now takes an option where you can specify a context where the
1466 given number will be matched. This way you have more control over who is allowed
1467 and it stops the people who blindly enter 10 digits.
1468 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
1469 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
1470 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
1471 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
1472 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
1473 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
1474 * The Dial() application no longer copies the language used by the caller to the callee's
1475 channel. If you desire for the caller's channel's language to be used for file playback
1476 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
1477 * SendImage() no longer hangs up the channel on error; instead, it sets the
1478 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
1479 'UNSUPPORTED'. This change makes SendImage() more consistent with other
1481 * Park has a new option, 's', which silences the announcement of the parking space number.
1482 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
1483 invalid input and will be assumed to mean that no timeout is desired.
1487 * Added DNS manager support to registrations for peers referencing peer entries.
1488 DNS manager runs in the background which allows DNS lookups to be run asynchronously
1489 as well as periodically updating the IP address. These properties allow for
1490 better performance as well as recovery in the event of an IP change.
1491 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
1492 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
1493 These changes also provide performance improvements for call setup and tear down.
1494 * Added ability to specify registration expiry time on a per registration basis in
1496 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
1498 * Added t38pt_usertpsource option. See sip.conf.sample for details.
1499 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
1500 * 'sip show peers' and 'sip show users' display their entries sorted in
1501 alphabetical order, as opposed to the order they were in, in the config
1503 * Videosupport now supports an additional option, "always", which always sets
1504 up video RTP ports, even on clients that don't support it. This helps with
1505 callfiles and certain transfers to ensure that if two video phones are
1506 connected, they will always share video feeds.
1510 * Existing DNS manager lookups extended to check for SRV records.
1511 * IAX2 encryption support has been improved to support periodic key rotation
1512 within a call for enhanced security. The option "keyrotate" has been
1513 provided to disable this functionality to preserve backwards compatibility
1514 with older versions of IAX2 that do not support key rotation.
1518 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
1519 data tree based on the given <path>.
1520 * New CLI command "data show providers" that will display all the registered
1522 * New CLI command, "config reload <file.conf>" which reloads any module that
1523 references that particular configuration file. Also added "config list"
1524 which shows which configuration files are in use.
1525 * New CLI commands, "pri show version" and "ss7 show version" that will
1526 display which version of libpri and libss7 are being used, respectively.
1527 A new API call was added so trunk will now have to be compiled against
1528 a versions of libpri and libss7 that have them or it will not know that
1529 these libraries exist.
1530 * The commands "core show globals", "core set global" and "core set chanvar" has
1531 been deprecated in favor of the more semanticly correct "dialplan show globals",
1532 "dialplan set chanvar" and "dialplan set global".
1533 * New CLI command "dialplan show chanvar" to list all variables associated
1534 with a given channel.
1538 * Addresses managed by DNS manager now can check to see if there is a DNS
1539 SRV record for a given domain and will use that hostname/port if present.
1541 AMI - The manager (TCP/TLS/HTTP)
1542 --------------------------------
1543 * The Status command now takes an optional list of variables to display
1544 along with channel status.
1545 * The QueueEntry event now also includes the channel's uniqueid
1549 * res_odbc no longer has a limit of 1023 total possible unshared connections,
1550 as some people were running into this limit. This limit has been increased
1555 * The TRANSFER queue log entry now includes the the caller's original
1556 position in the transferred-from queue.
1557 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
1558 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
1559 as well as an explanation about timeout options in general
1560 * Added a new option - C - for forcing the "answered elsewhere" flag on
1561 cancellation of calls in to members of the queue. This is to avoid the
1562 call to a member of a queue having the call listed as a "missed call".
1566 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
1567 adaptive capabilities. What this means in practical terms is that if your
1568 realtime table lacks critical fields, Asterisk will now emit warnings to
1569 that effect. Also, some of the realtime drivers have the ability (if
1570 configured) to automatically add those columns to the table with the
1571 correct type and length.
1575 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
1576 the 'setvar' option to cause a given audio file to be played upon completion
1577 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
1578 Skinny channels only.
1579 * You can now compile Asterisk against the Hoard Memory Allocator, see the
1580 Hoard page on the Asterisk wiki for more information:
1581 https://wiki.asterisk.org/wiki/x/pQBB
1582 * Config file variables may now be appended to, by using the '+=' append
1583 operator. This is most helpful when working with long SQL queries in
1584 func_odbc.conf, as the queries no longer need to be specified on a single
1586 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
1587 which will add a second to the billsec when the ending
1588 time is set, if the number in the microseconds field of the end time is
1589 greater than the number of microseconds in the answer time. This allows
1590 users to count the 'initiated' seconds in their billing records.
1592 ------------------------------------------------------------------------------
1593 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
1594 ------------------------------------------------------------------------------
1596 AMI - The manager (TCP/TLS/HTTP)
1597 --------------------------------
1598 * Manager has undergone a lot of changes, all of them documented
1599 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
1600 * Manager version has changed to 1.1
1601 * Added a new action 'CoreShowChannels' to list currently defined channels
1602 and some information about them.
1603 * Added a new action 'SIPshowregistry' to list SIP registrations.
1604 * Added TLS support for the manager interface and HTTP server
1605 * Added the URI redirect option for the built-in HTTP server
1606 * The output of CallerID in Manager events is now more consistent.
1607 CallerIDNum is used for number and CallerIDName for name.
1608 * Enable https support for builtin web server.
1609 See configs/http.conf.sample for details.
1610 * Added a new action, GetConfigJSON, which can return the contents of an
1611 Asterisk configuration file in JSON format. This is intended to help
1612 improve the performance of AJAX applications using the manager interface
1614 * SIP and IAX manager events now use "ChannelType" in all cases where we
1615 indicate channel driver. Previously, we used a mixture of "Channel"
1616 and "ChannelDriver" headers.
1617 * Added a "Bridge" action which allows you to bridge any two channels that
1618 are currently active on the system.
1619 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
1620 the voicemail users setup.
1621 * Added 'DBDel' and 'DBDelTree' manager commands.
1622 * cdr_manager now reports events via the "cdr" level, separating it from
1623 the very verbose "call" level.
1624 * Manager users are now stored in memory. If you change the manager account
1625 list (delete or add accounts) you need to reload manager.
1626 * Added Masquerade manager event for when a masquerade happens between
1628 * Added "manager reload" command for the CLI
1629 * Lots of commands that only provided information are now allowed under the
1630 Reporting privilege, instead of only under Call or System.
1631 * The IAX* commands now require either System or Reporting privilege, to
1632 mirror the privileges of the SIP* commands.
1633 * Added ability to retrieve list of categories in a config file.
1634 * Added ability to retrieve the content of a particular category.
1635 * Added ability to empty a context.
1636 * Created new action to create a new file.
1637 * Updated delete action to allow deletion by line number with respect to category.
1638 * Added new action insert to add new variable to category at specified line.
1639 * Updated action newcat to allow new category to be inserted in file above another
1641 * Added new event "JitterBufStats" in the IAX2 channel
1642 * Originate now requires the Originate privilege and, if you want to call out
1643 to a subshell, it requires the System privilege, as well. This was done to
1644 enhance manager security.
1645 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
1646 * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
1647 or manager show command Atxfer from the CLI
1648 * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
1649 details or manager show command IAXregistry from the CLI
1653 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
1654 state in the dialplan, as well as creating custom device states that are
1655 controllable from the dialplan.
1656 * Extend CALLERID() function with "pres" and "ton" parameters to
1657 fetch string representation of calling number presentation indicator
1658 and numeric representation of type of calling number value.
1659 * MailboxExists converted to dialplan function
1660 * A new option to Dial() for telling IP phones not to count the call
1661 as "missed" when dial times out and cancels.
1662 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
1663 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
1664 held for any given channel. Also, locks are automatically freed when a
1666 * Added HINT() dialplan function that allows retrieving hint information.
1667 Hints are mappings between extensions and devices for the sake of
1668 determining the state of an extension. This function can retrieve the list
1669 of devices or the name associated with a hint.
1670 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
1672 * Added SYSINFO() dialplan function which allows retrieval of system information
1673 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
1674 the existence of a dialplan target.
1675 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
1676 upper and lower case, respectively.
1677 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
1678 ID for the call (not the Asterisk call ID or unique ID), provided that the
1679 channel driver supports this. For SIP, you get the SIP call-ID for the
1680 bridged channel which you can store in the CDR with a custom field.
1684 * Added CLI permissions, config file: cli_permissions.conf
1685 default is to allow all commands for every local user/group.
1686 Also this new feature added three new CLI commands:
1687 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
1688 - cli reload permissions
1689 - cli show permissions
1690 * New CLI command "core show hint" (usage: core show hint <exten>)
1691 * New CLI command "core show settings"
1692 * Added 'core show channels count' CLI command.
1693 * Added the ability to set the core debug and verbose values on a per-file basis.
1694 * Added 'queue pause member' and 'queue unpause member' CLI commands
1695 * Ability to set process limits ("ulimit") without restarting Asterisk
1696 * Enhanced "agi debug" to print the channel name as a prefix to the debug
1697 output to make debugging on busy systems much easier.
1698 * New CLI commands "dialplan set extenpatternmatching true/false"
1699 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
1700 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
1701 listed in the startup_commands section of cli.conf will get executed.
1702 * Added a CLI command, "devstate change", which allows you to set custom device
1703 states from the func_devstate module that provides the DEVICE_STATE() function
1704 and handling of the "Custom:" devices.
1705 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
1706 sorted into the different possible callbacks, with the number of entries
1707 currently scheduled for each. Gives you a feel for how busy the sip channel
1709 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
1710 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
1711 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
1715 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
1716 option is enabled, Asterisk will watch for a CNG tone in the incoming audio
1717 for a received call. If it is detected, the channel will jump to the
1718 'fax' extension in the dialplan.
1719 * The default SIP useragent= identifier now includes the Asterisk version
1720 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
1721 If set, and the incoming request carries authentication info,
1722 the username to match in the users list is taken from the Digest header
1723 rather than from the From: field. This feature is considered experimental.
1724 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
1725 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
1726 * The "localmask" setting was removed in version 1.2 and the reminder about it
1727 being removed is now also removed.
1728 * A new option "busylevel" for setting a level of calls where asterisk reports
1729 a device as busy, to separate it from call-limit. This value is also added
1730 to the SIP_PEER dialplan function.
1731 * A new realtime family called "sipregs" is now supported to store SIP registration
1732 data. If this family is defined, "sippeers" will be used for configuration and
1733 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
1734 registration data, as before.
1735 * The SIPPEER function have new options for port address, call and pickup groups
1736 * Added support for T.140 realtime text in SIP/RTP
1737 * The "checkmwi" option has been removed from sip.conf, as it is no longer
1738 required due to the restructuring of how MWI is handled. See the descriptions
1739 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
1740 for more information.
1741 * Added rtpdest option to CHANNEL() dialplan function.
1742 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
1743 * SIP now adds a header to the CANCEL if the call was answered by another phone
1744 in the same dial command, or if the new c option in dial() is used.
1745 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
1746 states it is not needed. For phones, however, that do require it the "registertrying" option
1747 has been added so it can be enabled.
1748 * A new option called "callcounter" (global/peer/user level) enables call counters needed
1749 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
1750 used to enable this functionality).
1751 * New settings for timer T1 and timer B on a global level or per device. This makes it
1752 possible to force timeout faster on non-responsive SIP servers. These settings are
1753 considered advanced, so don't use them unless you have a problem.
1754 * Added a dial string option to be able to set the To: header in an INVITE to any
1756 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
1757 the qualify frequency.
1758 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
1759 were not properly torn down due to network or endpoint failures during an established
1761 * Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB
1762 and configs/sip.conf.sample for more information on how it is used.
1763 * Added a new configuration option "authfailureevents" that enables manager events when
1764 a peer can't authenticate properly.
1765 * Added DNS manager support to registrations for peers not referencing a peer entry.
1769 * Added the trunkmaxsize configuration option to chan_iax2.
1770 * Added the srvlookup option to iax.conf
1771 * Added support for OSP. The token is set and retrieved through the CHANNEL()
1774 XMPP Google Talk/Jingle changes
1775 -------------------------------
1776 * Added the bindaddr option to gtalk.conf.
1780 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
1781 * Proper codec support in chan_skinny.
1782 * Added settings for IP and Ethernet QoS requests
1786 * Added separate settings for media QoS in mgcp.conf
1788 Console Channel Driver changes
1789 ------------------------------
1790 * Added experimental support for video send & receive to chan_oss.
1791 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
1794 Phone channel changes (chan_phone)
1795 ----------------------------------
1796 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
1798 H.323 channel Changes
1799 ---------------------
1800 * H323 remote hold notification support added (by NOTIFY message
1801 and/or H.450 supplementary service)
1803 Local channel changes
1804 ---------------------
1805 * The device state functionality in the Local channel driver has been updated
1806 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
1807 to just UNKNOWN if the extension exists.
1808 * Added jitterbuffer support for chan_local. This allows you to use the
1809 generic jitterbuffer on incoming calls going to Asterisk applications.
1810 For example, this would allow you to use a jitterbuffer for an incoming
1811 SIP call to Voicemail by putting a Local channel in the middle. This
1812 feature is enabled by using the 'j' option in the Dial string to the Local
1813 channel in conjunction with the existing 'n' option for local channels.
1814 * A 'b' option has been added which causes chan_local to return the actual channel
1815 that is behind it when queried. This is useful for transfer scenarios as the
1816 actual channel will be transferred, not the Local channel.
1818 Agent channel changes
1819 ----------------------
1820 * The ackcall and endcall options are now supplemented with options acceptdtmf
1821 and enddtmf. These allow for the DTMF keypress to be configurable. The options
1822 default to their old hard-coded values ('#' and '*' respectively) so this should
1823 not break any existing agent installations.
1825 DAHDI channel driver (chan_dahdi) Changes
1826 ----------------------------------------
1827 * SS7 support (via libss7 library)
1828 * In India, some carriers transmit CID via dtmf. Some code has been added
1829 that will handle some situations. The cidstart=polarity_IN choice has been added for
1830 those carriers that transmit CID via dtmf after a polarity change.
1831 * CID matching information is now shown when doing 'dialplan show'.
1832 * Added dahdi show version CLI command.
1833 * Added setvar support to chan_dahdi.conf channel entries.
1834 * Added two new options: mwimonitor and mwimonitornotify. These options allow
1835 you to enable MWI monitoring on FXO lines. When the MWI state changes,
1836 the script specified in the mwimonitornotify option is executed. An internal
1837 event indicating the new state of the mailbox is also generated, so that
1838 the normal MWI facilities in Asterisk work as usual.
1839 * Added signalling type 'auto', which attempts to use the same signalling type
1840 for a channel as configured in DAHDI. This is primarily designed for analog
1841 ports, but will also work for digital ports that are configured for FXS or FXO
1842 signalling types. This mode is also the default now, so if your chan_dahdi.conf
1843 does not specify signalling for a channel (which is unlikely as the sample
1844 configuration file has always recommended specifying it for every channel) then
1845 the 'auto' mode will be used for that channel if possible.
1846 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
1847 state for a channel; also ensured that the DNDState Manager event is
1848 emitted no matter how the DND state is set or cleared.
1852 * Added a new channel driver, chan_unistim. See the Asterisk wiki at
1853 https://wiki.asterisk.org/wiki/x/vgsiAQ and configs/unistim.conf.sample
1854 for details. This new channel driver allows you to use Nortel i2002,
1855 i2004, and i2050 phones with Asterisk.
1856 * Added a new channel driver, chan_console, which uses portaudio as a cross
1857 platform audio interface. It was written as a channel driver that would
1858 work with Mac CoreAudio, but portaudio supports a number of other audio
1859 interfaces, as well. Note that this channel driver requires v19 or higher
1860 of portaudio; older versions have a different API.
1864 * Added the ability to specify arguments to the Dial application when using
1865 the DUNDi switch in the dialplan.
1866 * Added the ability to set weights for responses dynamically. This can be
1867 done using a global variable or a dialplan function. Using the SHELL()
1868 function would allow you to have an external script set the weight for
1870 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
1871 functions will allow you to initiate a DUNDi query from the dialplan,
1872 find out how many results there are, and access each one.
1873 * Added the ability to specifiy a port for a dundi peer.
1877 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
1878 functions will allow you to initiate an ENUM lookup from the dialplan,
1879 and Asterisk will cache the results. ENUMRESULT can be used to access
1880 the results without doing multiple DNS queries.
1884 * Added the ability to customize which sound files are used for some of the
1885 prompts within the Voicemail application by changing them in voicemail.conf
1886 * Added the ability for the "voicemail show users" CLI command to show users
1887 configured by the dynamic realtime configuration method.
1888 * MWI (Message Waiting Indication) handling has been significantly
1889 restructured internally to Asterisk. It is now totally event based
1890 instead of polling based. The voicemail application will notify other
1891 modules that have subscribed to MWI events when something in the mailbox
1893 This also means that if any other entity outside of Asterisk is changing
1894 the contents of mailboxes, then the voicemail application still needs to
1895 poll for changes. Examples of situations that would require this option
1896 are web interfaces to voicemail or an email client in the case of using
1897 IMAP storage. So, two new options have been added to voicemail.conf
1898 to account for this: "pollmailboxes" and "pollfreq". See the sample
1899 configuration file for details.
1900 * Added "tw" language support
1901 * Added support for storage of greetings using an IMAP server
1902 * Added ability to customize forward, reverse, stop, and pause keys for message playback
1903 * SMDI is now enabled in voicemail using the smdienable option.
1904 * A "lockmode" option has been added to asterisk.conf to configure the file
1905 locking method used for voicemail, and potentially other things in the
1906 future. The default is the old behavior, lockfile. However, there is a
1907 new method, "flock", that uses a different method for situations where the
1908 lockfile will not work, such as on SMB/CIFS mounts.
1909 * Added the ability to backup deleted messages, to ease recovery in the case
1910 that a user accidentally deletes a message, and discovers that they need it.
1911 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
1912 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
1913 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
1914 voicemail boxes. The SMDI interface can also poll for MWI changes when some
1915 outside entity is modifying the state of the mailbox (such as IMAP storage or
1916 a web interface of some kind).
1917 * Added the support for marking messages as "urgent." There are two methods to accomplish
1918 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
1919 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
1920 the message as urgent after he has recorded a voicemail by following the voice instructions.
1921 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
1926 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
1927 used across multiple queues.
1928 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
1929 setqueueentryvar options for each queue, see queues.conf.sample for details.
1930 * Added keepstats option to queues.conf which will keep queue
1931 statistics during a reload.
1932 * setinterfacevar option in queues.conf also now sets a variable
1933 called MEMBERNAME which contains the member's name.
1934 * Added 'Strategy' field to manager event QueueParams which represents
1935 the queue strategy in use.
1936 * Added option to run macro when a queue member is connected to a caller,
1937 see queues.conf.sample for details.
1938 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
1939 does not count paused queue members as unavailable.
1940 * Added min-announce-frequency option to queues.conf which allows you to control the
1941 minimum amount of time between queue announcements for use when the caller's queue
1942 position changes frequently.
1943 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
1945 * Added ability for non-realtime queues to have realtime members
1946 * Added the "linear" strategy to queues.
1947 * Added the "wrandom" strategy to queues.
1948 * Added new channel variable QUEUE_MIN_PENALTY
1949 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
1950 rules in queuerules.conf. See configs/queuerules.conf.sample for details
1951 * Added a new parameter for member definition, called state_interface. This may be
1952 used so that a member may be called via one interface but have a different interface's
1953 device state reported.
1954 * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
1955 "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
1956 "manager show command QueueReset."
1957 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
1958 specified by the periodic-announce option, then one will be chosen randomly when it is time
1959 to play a periodic announcment
1960 * New configuration options: announce-position now takes two more values in addition to "yes" and
1961 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
1962 announce-position-limit. By setting announce-position to "limit" callers will only have their
1963 position announced if their position is less than what is specified by announce-position-limit.
1964 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
1965 will be told that their are more than announce-position-limit callers waiting.
1966 * Two new queue log events have been added. An ADDMEMBER event will be logged
1967 when a realtime queue member is added and a REMOVEMEMBER event will be logged
1968 when a realtime queue member is removed. Since there is no calling channel associated
1969 with these events, the string "REALTIME" is placed where the channel's unique id
1970 is typically placed.
1971 * The configuration method for the "joinempty" and "leavewhenempty" options has
1972 changed to a comma-separated list of methods of determining member availability
1973 instead of vague terms such as "yes," "loose," "no," and "strict." These old four
1974 values are still accepted for backwards-compatibility, though.
1975 * The average talktime is now calculated on queues. This information is reported via the
1976 CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
1977 and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
1982 * The 'o' option to provide an optimization has been removed and its functionality
1983 has been enabled by default.
1984 * When a conference is created, the UNIQUEID of the channel that caused it to be
1985 created is stored. Then, every channel that joins the conference will have the
1986 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
1987 callers that come and go from long standing conferences.
1988 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
1989 except it does operations on a channel by name, instead of number in a conference.
1990 This is a very useful feature in combination with the 'X' option to ChanSpy.
1991 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
1993 * Added new RealTime functionality to provide support for scheduled conferencing.
1994 This includes optional messages to the caller if they attempt to join before
1995 the schedule start time, or to allow the caller to join the conference early.
1996 Also included is optional support for limiting the number of callers per
1997 RealTime conference.
1998 * Added the S() and L() options to the MeetMe application. These are pretty
1999 much identical to the S() and L() options to Dial(). They let you set
2000 timeouts for the conference, as well as have warning sounds played to
2001 let the caller know how much time is left, and when it is running out.
2002 * Added the ability to do "meetme concise" with the "meetme" CLI command.
2003 This extends the concise capabilities of this CLI command to include
2004 listing all conferences, instead of an addition to the other sub commands
2005 for the "meetme" command.
2006 * Added the ability to specify the music on hold class used to play into the
2007 conference when there is only one member and the M option is used.
2008 * Added MEETME_INFO dialplan function which provides a way to query
2009 various properties of a Meetme conference.
2010 * Added new admin features: *81: Roll call, *82: eject all, *83: mute all,
2011 and *84: record in-conf
2013 Other Dialplan Application Changes
2014 ----------------------------------
2015 * Argument support for Gosub application
2016 * From the to-do lists: straighten out the app timeout args:
2017 Wait() app now really does 0.3 seconds- was truncating arg to an int.
2018 WaitExten() same as Wait().
2019 Congestion() - Now takes floating pt. argument.
2020 Busy() - now takes floating pt. argument.
2021 Read() - timeout now can be floating pt.
2022 WaitForRing() now takes floating pt timeout arg.
2023 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
2024 * Added 's' option to Page application.
2025 * Added an optional timeout argument to the Page application.
2026 * Added 'E', 'V', and 'P' commands to ExternalIVR.
2027 * Added 'o' and 'X' options to Chanspy.
2028 * Added a new dialplan application, Bridge, which allows you to bridge the
2029 calling channel to any other active channel on the system.
2030 * Added the ability to specify a music on hold class to play instead of ringing
2031 for the SLATrunk application.
2032 * The Read application no longer exits the dialplan on error. Instead, it sets
2033 READSTATUS to ERROR, which you can catch and handle separately.
2034 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
2035 of asking for verification of each name, one at a time.
2036 * Privacy() no longer uses privacy.conf, as all options are specifyable as
2037 direct options to the app.
2038 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
2040 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
2041 * The ChannelRedirect application no longer exits the dialplan if the given channel
2042 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
2043 or NOCHANNEL if the given channel was not found.
2044 * The silencethreshold setting that was previously configurable in multiple
2045 applications is now settable globally via dsp.conf.
2047 Music On Hold Changes
2048 ---------------------
2049 * A new option, "digit", has been added for music on hold classes in
2050 musiconhold.conf. If this is set for a music on hold class, a caller
2051 listening to music on hold can press this digit to switch to listening
2052 to this music on hold class.
2053 * Support for realtime music on hold has been added.
2054 * In conjunction with the realtime music on hold, a general section has
2055 been added to musiconhold.conf, its sole variable is cachertclasses. If this
2056 is set, then music on hold classes found in realtime will be cached in memory.
2060 * AEL upgraded to use the Gosub with Arguments instead
2061 of Macro application, to hopefully reduce the problems
2062 seen with the artificially low stack ceiling that
2063 Macro bumps into. Macros can only call other Macros
2064 to a depth of 7. Tests run using gosub, show depths
2065 limited only by virtual memory. A small test demonstrated
2066 recursive call depths of 100,000 without problems.
2067 -- in addition to this, all apps that allowed a macro
2068 to be called, as in Dial, queues, etc, are now allowing
2069 a gosub call in similar fashion.
2070 * AEL now generates LOCAL(argname) declarations when it
2071 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
2072 etc. That makes the arguments local in scope. The user
2073 can define their own local variables in macros, now,
2074 by saying "local myvar=someval;" or using Set() in this
2075 fashion: Set(LOCAL(myvar)=someval); ("local" is now
2077 * utils/conf2ael introduced. Will convert an extensions.conf
2078 file into extensions.ael. Very crude and unfinished, but
2079 will be improved as time goes by. Should be useful for a
2080 first pass at conversion.
2081 * aelparse will now read extensions.conf to see if a referenced
2082 macro or context is there before issueing a warning.
2083 * AEL parser sets a local channel variable ~~EXTEN~~, to
2084 preserve the value of ${EXTEN} thru switch statements.
2085 * New operator in $[...] expressions: the ~~ operator serves
2086 as a concatenation operator. AT THE MOMENT, it is really only
2087 necessary and useful in AEL, especially in if() expressions.
2088 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
2089 any enclosing double-quotes, and evaluate to the value of a
2090 concatenated with the value of b. For example if a is set to
2091 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
2092 evaluate to xyzabc .
2095 Call Features (res_features) Changes
2096 ------------------------------------
2097 * Added the parkedcalltransfers option to features.conf
2098 * Added parkedcallparking option to control one touch parking w/ parking
2100 * Added parkedcallhangup option to control disconnect feature w/ parking
2102 * Added parkedcallrecording option to control one-touch record w/ parking
2104 * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
2105 parkedcalltransfers option support for multiple parking lots.
2106 * Added BRIDGE_FEATURES variable to set available features for a channel
2107 * The built-in method for doing attended transfers has been updated to
2108 include some new options that allow you to have the transferee sent
2109 back to the person that did the transfer if the transfer is not successful.
2110 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
2111 in features.conf.sample.
2112 * Added support for configuring named groups of custom call features in
2113 features.conf. This means that features can be written a single time, and
2114 then mapped into groups of features for different key mappings or easier
2116 * Updated the ParkedCall application to allow you to not specify a parking
2117 extension. If you don't specify a parking space to pick up, it will grab
2118 the first one available.
2119 * Added cli command 'features reload' to reload call features from features.conf
2120 * Moved into core asterisk binary.
2121 * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
2122 * Added the ability for custom parking lots to be configured with their own
2123 parking extension with the parkext option.
2125 Language Support Changes
2126 ------------------------
2127 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
2128 * Added support for the Hungarian language for saying numbers, dates, and times.
2132 * Added SPEECH commands for speech recognition. A complete listing can be found
2134 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
2135 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
2136 does not behave as expected; the native command needs to be used, instead.
2137 * Added the ability to perform SRV lookups on fast AGI calls. To use this
2138 feature, simply use hagi: instead of agi: as the protocol portion
2139 of the URI parameter to the AGI function call in your dial plan. Also note
2140 that specifying a port number in the AGI URI will disable SRV lookups,
2141 even if you use the hagi: protocol.
2142 * No longer support MSG_OOB flag on HANGUP.
2146 * Added rotatestrategy option to logger.conf, along with two new options:
2147 "timestamp" which will use the time to name the logger files instead of
2148 sequence number; and "rotate", which rotates the names of the log files,
2149 similar to the way syslog rotates files.
2150 * Added exec_after_rotate option to logger.conf, which allows a system
2151 command to be run after rotation. This is primarily useful with
2152 rotatestrategy=rotate, to allow a limit on the number of log files kept
2153 and to ensure that the oldest log file gets deleted.
2154 * Added realtime support for the queue log
2158 * The cdr_manager module has a [mappings] feature, like cdr_custom,
2159 to add fields to the manager event from the CDR variables.
2160 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
2161 backend database CDR table. Specifically, additional, non-standard
2162 columns are supported, merely by setting the corresponding CDR variable in
2163 your dialplan. In addition, you may alias any column to another name (for
2164 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
2165 simply "alias src => ANI" in the configuration file). Records may be
2166 posted to more than one backend, simply by specifying multiple categories
2167 in the configuration file. And finally, you may filter which CDRs get
2168 posted to each backend, by specifying a filter (which the record must
2169 match) for the particular category. Filters are additive (meaning all
2170 rules must match to post that CDR).
2171 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
2172 module. Specifically, you may add additional columns into the table and
2173 they will be set, if you set the corresponding CDR variable name. Also,
2174 if you omit columns in your database table, they will be silently skipped
2175 (but a record will still be inserted, based on what columns remain). Note
2176 that the other two features from cdr_adaptive_odbc (alias and filter) are
2177 not currently supported.
2178 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
2179 has been disabled using the NoCDR application.
2181 Miscellaneous New Modules
2182 -------------------------
2183 * Added a new CDR module, cdr_sqlite3_custom.
2184 * Added a new realtime configuration module, res_config_sqlite
2185 * Added a new codec translation module, codec_resample, which re-samples
2186 signed linear audio between 8 kHz and 16 kHz to help support wideband
2188 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
2189 based on configuration templates that use Asterisk dialplan function and
2190 variable substitution. It should be possible to create phone profiles and
2191 templates that work for the majority of phones provisioned over http. It
2192 is currently only intended to provision a single user account per phone.
2193 An example profile and set of templates for Polycom phones is provided.
2194 NOTE: Polycom firmware is not included, but should be placed in
2195 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
2196 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
2197 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
2198 provided; there is a JACK() application, and a JACK_HOOK() function. Both
2199 interfaces create an input and output JACK port. The application makes
2200 these ports the endpoint of the call. The audio coming from the channel
2201 goes out the output port and whatever comes back in on the input port is
2202 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
2203 audiohook on the channel. This lets you run the audio coming from a
2204 channel through JACK, and whatever comes back in is what gets forwarded
2205 on as the channel's audio. This is very useful for building custom
2206 vocoders or doing recording or analysis of the channel's audio in another
2208 * Added a new module, res_config_curl, which permits using a HTTP POST url
2209 to retrieve, create, update, and delete realtime information from a remote
2210 web server. Note that this module requires func_curl.so to be loaded for
2211 backend functionality.
2212 * Added a new module, res_config_ldap, which permits the use of an LDAP
2213 server for realtime data access.
2214 * Added support for writing and running your dialplan in lua using the pbx_lua
2215 module. See configs/extensions.lua.sample for examples of how to do this.
2219 * Ability to use libcap to set high ToS bits when non-root
2220 on Linux. If configure is unable to find libcap then you
2221 can use --with-cap to specify the path.
2222 * Added maxfiles option to options section of asterisk.conf which allows you to specify
2223 what Asterisk should set as the maximum number of open files when it loads.
2224 * Added the jittertargetextra configuration option.
2225 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
2226 configuration files for the IP channel drivers. The new option is "cos".
2227 This information is also documented on the Asterisk wiki at
2228 https://wiki.asterisk.org/wiki/x/EYBG
2229 * When originating a call using AMI or pbx_spool that fails the reason for failure
2230 will now be available in the failed extension using the REASON dialplan variable.
2231 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
2232 It allows you to configure a prefix for auto-monitor recordings.
2233 * A new extension pattern matching algorithm, based on a trie, is introduced
2234 here, that could noticeably speed up mid-sized to large dialplans.
2235 It is NOT used by default, as duplicating the behaviour of the old pattern
2236 matcher is still under development. A config file option, in extensions.conf,
2237 in the [general] section, called "extenpatternmatchingnew", is by default
2238 set to false; setting that to true will force the use of the new algorithm.
2239 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
2240 be used to switch the algorithms at run time.
2241 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
2242 specifying which socket to use to connect to the running Asterisk daemon
2244 * Performance enhancements to the sched facility, which is used in
2245 the channel drivers, etc. Added hashtabs and doubly-linked lists
2246 to speed up deletion; start at the beginning or end of list to
2248 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
2249 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
2250 Added regression tests to the tests/ dir, also.
2251 * Added a refcount trace feature to astobj2 for those trying to balance
2252 object creation, deletion; work, play; space and time. See the
2253 notes in astobj2.h. Also, see utils/refcounter as well, as a
2254 quick way to find unbalanced refcounts in what could be a sea
2255 of objects that were balanced.
2256 * Added logging to 'make update' command. See update.log
2257 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
2258 do not come from the remote party.
2259 * Added the 'n' option to the SpeechBackground application to tell it to not
2260 answer the channel if it has not already been answered.
2261 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
2262 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
2264 * iLBC source code no longer included (see UPGRADE.txt for details)
2265 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
2266 deadlock is detected, a backtrace of the stack which led to the lock calls
2267 will be output to the CLI.
2268 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
2269 the "core show locks" CLI command will give lock information output as well
2270 as a backtrace of the stack which led to the lock calls.
2271 * users.conf now sports an optional alternateexts property, which permits
2272 allocation of additional extensions which will reach the specified user.
2273 * A new option for the configure script, --enable-internal-poll, has been added
2274 for use with systems which may have a buggy implementation of the poll system
2275 call. If you notice odd behavior such as the CLI being unresponsive on remote
2276 consoles, you may want to try using this option. This option is enabled by default
2277 on Darwin systems since it is known that the Darwin poll() implementation has
2281 --------------------
2282 * In addition to timing from DAHDI, there is a new timing module called
2283 res_timing_timerfd. In order to use this, you must be running Linux with
2284 a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
2285 script will be able to tell if you have the requirements. From menuselect, select
2286 res_timing_timerfd from the Resource Modules menu.
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