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
61 * mixmonitor list <channel> command will now show MixMonitor ID, and the filenames
62 of all running mixmonitors on a channel.
63 * The debuglevel of "pri set debug" is now a bitmask ranging from 0 to 15 if
64 numeric instead of 0, 1, or 2.
65 * "stun show status" will show a table describing how the STUN client is behaving.
69 * Added menu action admin_toggle_mute_participants. This will mute / unmute
70 all non-admin participants on a conference. The confbridge configuration file
71 also allows for the default sounds played to all conference users when this
72 occurs to be overriden using sound_participants_unmuted and sound_participants_muted.
73 * Added menu action participant_count. This will playback the number of current
74 participants in a conference.
75 * Added announcement configuration option to user profile. If set the sound file will
76 be played to the user, and only the user, upon joining the conference bridge.
80 * Addition of the VM_INFO function - see Dialplan function changes
81 * The imapserver, imapport, and imapflags configuration options can now be
82 overriden on a user by user basis.
86 * Asterisk will no longer substitute CID number for CID name into display
87 name field if CID number exists without a CID name. This change improves
88 compatibility with certain device features such as Avaya IP500's directory
90 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
91 created using that setting to not be removed during SIP reload.
92 * Add support to realtime for the 'callbackextension' option
93 * When multiple peers exist with the same address, but differing
94 callbackextension options, incoming requests that are matched by address
95 will be matched to the peer with the matching callbackextension if it is
97 * NAT settings are now a combinable list of options. The equivalent of the
98 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
99 * Two new NAT options, auto_force_rport and auto_comedia, have been added
100 which set the force_rport and comedia options automatically if Asterisk
101 detects that an incoming SIP request crossed a NAT after being sent by
103 * Adds an option send_diversion which can be disabled to prevent
104 diversion headers from automatically being added to invites.
105 * Add support for lightweight NAT keepalive. If enabled a blank packet will
106 be sent to the remote host at a given interval to keep the NAT mapping open.
107 This can be enabled using the keepalive configuration option.
108 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
113 * Added a manager event "LocalBridge" for local channel call bridges between
114 the two pseudo-channels created.
118 * Added dialtone_detect option for analog ports to disconnect incoming
119 calls when dialtone is detected.
121 ------------------------------------------------------------------------------
122 --- Functionality changes since Asterisk 10.4.0 ------------------------------
123 ------------------------------------------------------------------------------
127 * The optimization portion of the build system has been reworked to avoid
128 broken builds on certain architectures. All architecture-specific
129 optimization has been removed in favor of using -march=native to allow gcc
130 to detect the environment in which it is running when possible. This can
131 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
133 ------------------------------------------------------------------------------
134 --- Functionality changes since Asterisk 10.3.0 ------------------------------
135 ------------------------------------------------------------------------------
139 * Added ability to use multiple lines on phone, so for one device in
140 configuration multiple lines can be defined, it allows to have multiple calls
141 on one phone, callwaiting and switching between calls.
142 * Added option 'sharpdial' allowing end dialing by pressing # key
143 * Added option 'interdigit_timer' for controll phone dial timeout
144 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
145 * Added global 'debug' option, that enables debug in channel driver
146 * Added ability for translation on-screen menu to multiple languages. Tested on
147 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
148 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
150 * In addition to English added French and Russian languages for on-screen menus
151 * Reworked dialing number input: added dialing by timeout, immediate dial on
152 on dialplan compare, phone number length now not limited by screen size
153 * Added ability for pickup a call using features.conf defined value and
158 * Codec lists may now be modified by the '!' character, to allow succinct
159 specification of a list of codecs allowed and disallowed, without the
160 requirement to use two different keywords. For example, to specify all
161 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
163 Music On Hold Changes
164 ---------------------
165 * Added 'announcement' option which will play at the start of MOH and between
166 songs in modes of MOH that can detect transitions between songs (eg.
171 * Added queue options autopausebusy and autopauseunavail for automatically
172 pausing a queue member when their device reports busy or congestion.
173 * The 'ignorebusy' option for queue members has been deprecated in favor of
174 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
175 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
176 per interface basis. Individual ringinuse values can now be set in
177 queues.conf via an argument to member definitions. Lastly, the queue
178 'ringinuse' setting now only determines defaults for the per member
179 'ringinuse' setting and does not override per member settings like it does
184 * When voicemail plays a message's envelope with saycid set to yes, when reaching
185 the caller id field it will play a recording of a file with the same base name
186 as the sender's callerid if there is a similarly named file in
187 <astspooldir>/recordings/callerids/
191 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
192 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
193 changed arguments to SayUnixTime so that every option is truly optional even
194 when using multiple options (so that j option could be used without having to
195 manually specify timezone and format) There are other beneftis eg. format can
196 now be used without specifying time zone as well.
197 * Added 'F()' option to Queue and Bridge. Similar to the dial option, these can
198 be supplied with arguments indicating where the callee should go after the caller
199 is hung up, or without options specified, the priority after the Queue/Bridge
201 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
202 channels respectively before the callee channels are called.
206 * New per parking lot options: comebackcontext and comebackdialtime. See
207 configs/features.conf.sample for more details.
209 * Channel variable PARKER is now set when comebacktoorigin is disabled in
212 * MixMonitor hooks now have IDs associated with them which can be used to assign
213 a target to StopMixMonitor. Use of MixMonitor's i(variable) option will allow
214 storage of the MixMontior ID in a channel variable. StopMixmonitor now accepts
215 that ID as an argument.
217 CDR postgresql driver changes
218 -----------------------------
219 * Added command "cdr show pgsql status" to check connection status
221 AMI (Asterisk Manager Interface) changes
222 ----------------------------------------
223 * Originate now generates an error response if the extension given
224 is not found in the dialplan
226 * MixMonitor will now show IDs associated with the mixmonitor upon creating them
227 if the i(variable) option is used. StopMixMonitor will accept MixMonitorID as
228 on option to close specific MixMonitors.
230 * The SIPshowpeer manager action response field "SIP-Forcerport" has been updated
231 to include information about peers configured with nat=auto_force_rport by
232 returning "A" if auto_force_rport is set and nat is detected, and "a" if it is
233 set and nat is not detected. "Y" and "N" are still returned if auto_force_rport
236 * Hangup now can take a regular expression as the Channel option. If you want
237 to hangup multiple channels, use /regex/ as the Channel option. Existing
238 behavior to hanging up a single channel is unchanged, but if you pass a regex,
239 the manager will send you a list of channels back that were hung up.
241 * Support for IPv6 addresses has been added.
243 * AMI Events can now be documented in the Asterisk source. Two new CLI
244 commands have been added to display information about AMI events at run time:
245 manager show events, which shows a list of all known and documented AMI
246 events, and manager show event [event name], which shows detail information
247 about a specific AMI event. Note that AMI event documentation is only
248 generated when Asterisk is compiled using 'make full'.
252 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
253 control of faxdetect.
257 * Allow the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to be
258 used within the dynamic weight attribute when specifying a mapping.
262 * Addition of the VM_INFO function that can be used to retrieve voicemail
263 user information, such as the email address and full name.
264 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
266 * The REDIRECTING function now supports the redirecting original party id
268 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
269 lets you set some of the configuration options from the [general] section
270 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
271 the key sequence used to activate built-in features, such as blindxfer,
272 and automon. See the built-in documentation for details.
276 * A new option, 'I' has been added to app_followme.
277 By setting this option, Asterisk will not update the caller with
278 connected line changes when they occur. This is similar to app_dial
280 * The 'N' option is now ignored if the call is already answered.
281 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
282 and caller channels respectively before the callee channels are called.
286 * A new option, 'probation' has been added to rtp.conf
287 RTP in strictrtp mode can now require more than 1 packet to exit learning
288 mode with a new source (and by default requires 4). The probation option
289 allows the user to change the required number of packets in sequence to any
290 desired value. Use a value of 1 to essentially restore the old behavior.
291 Also, with strictrtp on, Asterisk will now drop all packets until learning
292 mode has successfully exited. These changes are based on how pjmedia handles
293 media sources and source changes.
297 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
298 instead of simply the uri. This is the format that MessageSend() can use
299 in the from parameter for outgoing SIP messages.
303 * A new module, res_corosync, has been introduced. This module uses the
304 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
305 of Asterisk servers to both Message Waiting Indication (MWI) and/or
306 Device State (presence) information. This module is very similar to, and
307 is a replacement for the res_ais module that was in previous releases of
312 * A new channel variable, AGIEXITONHANGUP, has been added which allows
313 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
314 AGI application would exit immediately after a channel hangup is detected.
315 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
316 are resolved and each address is attempted in turn until one succeeds or
321 * Direct media functionality has been added.
322 Options in config are: directmedia (directrtp) and directrtpsetup (earlydirect)
326 * A new channel driver named chan_motif has been added which provides support for
327 Google Talk and Jingle in a single channel driver. This new channel driver includes
328 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
329 hold, unhold, and ringing notification. It is also compliant with the current Jingle
330 specification, current Google Jingle specification, and the original Google Talk
333 ------------------------------------------------------------------------------
334 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
335 ------------------------------------------------------------------------------
339 * Asterisk now has protocol independent support for processing text messages
340 outside of a call. Messages are routed through the Asterisk dialplan.
341 SIP MESSAGE and XMPP are currently supported. There are options in
342 jabber.conf and sip.conf to allow enabling these features.
343 -> jabber.conf: see the "sendtodialplan" and "context" options.
344 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
345 and "outofcall_message_context" options.
346 The MESSAGE() dialplan function and MessageSend() application have been
347 added to go along with this functionality. More detailed usage information
348 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
349 * If real-time text support (T.140) is negotiated, it will be preferred for
350 sending text via the SendText application. For example, via SIP, messages
351 that were once sent via the SIP MESSAGE request would be sent via RTP if
352 T.140 text is negotiated for a call.
356 * parkedmusicclass can now be set for non-default parking lots.
358 Asterisk Manager Interface
359 --------------------------
360 * PeerStatus now includes Address and Port.
361 * Added Hold events for when the remote party puts the call on and off hold
362 for chan_dahdi ISDN channels.
363 * Added new action MeetmeListRooms to list active conferences (shows same
364 data as "meetme list" at the CLI).
365 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
366 Description field that is set by 'description' in the channel configuration
368 * Added Uniqueid header to UserEvent.
369 * Added new action FilterAdd to control event filters for the current session.
370 This requires the system permission and uses the same filter syntax as
371 filters that can be defined in manager.conf
372 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
373 versions had some instances of the event converted, but others were left
374 as-is. All Unlink events should now be converted to Bridge events. The AMI
375 protocol version number was incremented to 1.2 as a result of this change.
378 --------------------------
379 * The HTTP Server can bind to IPv6 addresses.
382 --------------------------
383 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
384 with busydetect. usage example: busypattern=200,200,200,600
387 --------------------------
388 * New 'gtalk show settings' command showing the current settings loaded from
390 * The 'logger reload' command now supports an optional argument, specifying an
391 alternate configuration file to use.
392 * 'dialplan add extension' command will now automatically create a context if
393 the specified context does not exist with a message indicated it did so.
394 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
395 Description field which can be populated with 'description' in the channel
396 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
399 --------------------------
400 * The filter option in cdr_adaptive_odbc now supports negating the argument,
401 thus allowing records which do NOT match the specified filter.
402 * Added ability to log CONGESTION calls to CDR
405 --------------------------
406 * Ability to define custom SILK formats in codecs.conf.
407 * Addition of speex32 audio format with translation.
408 * CELT codec pass-through support and ability to define
409 custom CELT formats in codecs.conf.
410 * Ability to read raw signed linear files with sample rates
411 ranging from 8khz - 192khz. The new file extensions introduced
412 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
413 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
414 Skinny, H.323, etc) can still only support the following codecs:
415 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
416 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
417 Video: h261, h263, h263p, h264, mpeg4
422 --------------------------
423 * New highly optimized and customizable ConfBridge application capable of
424 mixing audio at sample rates ranging from 8khz-96khz.
425 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
426 and bridge profiles on a channel.
427 * CONFBRIDGE_INFO dialplan function capable of retrieving information
428 about a conference such as locked status and number of parties, admins,
430 * Addition of video_mode option in confbridge.conf for adding video support
431 into a bridge profile.
432 * Addition of the follow_talker video_mode in confbridge.conf. This video
433 mode dynamically switches the video feed to always display the loudest talker
434 supplying video in the conference.
438 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
439 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
440 variables from asterisk.conf.
444 * Addition of the JITTERBUFFER dialplan function. This function allows
445 for jitterbuffering to occur on the read side of a channel. By using
446 this function conference applications such as ConfBridge and MeetMe can
447 have the rx streams jitterbuffered before conference mixing occurs.
448 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
450 * Added STRREPLACE function. This function let's the user search a variable
451 for a given string to replace with another string as many times as the
452 user specifies or just throughout the whole string.
453 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
454 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
455 * Added extensions to chan_ooh323 in function CHANNEL()
457 libpri channel driver (chan_dahdi) DAHDI changes
458 --------------------------
459 * Added moh_signaling option to specify what to do when the channel's bridged
460 peer puts the ISDN channel on hold.
461 * Added display_send and display_receive options to control how the display ie
462 is handled. To send display text from the dialplan use the SendText()
463 application when the option is enabled.
464 * Added mcid_send option to allow sending a MCID request on a span.
467 --------------------------
468 * Added setvar option to calendar.conf to allow setting channel variables on
469 notification channels.
470 * Added "calendar show types" CLI command to list registered calendar
474 --------------------------
475 * Added two new options, r and t with file name arguments to record
476 single direction (unmixed) audio recording separate from the bidirectional
477 (mixed) recording. The mixed file name argument is optional now as long
478 as at least one recording option is used.
481 --------------------------
482 * Added a new option, l, which will disable local call optimization for
483 channels involved with the FollowMe thread. Use this option to improve
484 compatability for a FollowMe call with certain dialplan apps, options, and
488 --------------------------
489 * Added option "k" that will automatically close the conference when there's
490 only one person left when a user exits the conference.
493 --------------------------
494 * cel_pgsql now supports the 'extra' column for data added using the
495 CELGenUserEvent() application.
498 --------------------------
499 * Support for defining hints has been added to pbx_lua. See the 'hints' table
500 in the sample extensions.lua file for syntax details.
501 * Applications that perform jumps in the dialplan such as Goto will now
502 execute properly. When pbx_lua detects that the context, extension, or
503 priority we are executing on has changed it will immediately return control
504 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
505 the priority after the currently executing priority.
506 * An autoservice is now started by default for pbx_lua channels. It can be
507 stopped and restarted using the autoservice_stop() and autoservice_start()
511 --------------------------
512 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
513 into a FAXStatus event with an 'Operation' header that will be either
514 'send', 'receive', and 'gateway'.
515 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
516 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
517 feature will handle converting a fax call between an audio T.30 fax terminal
518 and an IFP T.38 fax terminal.
522 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
523 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
524 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
528 * Added general option negative_penalty_invalid default off. when set
529 members are seen as invalid/logged out when there penalty is negative.
530 for realtime members when set remove from queue will set penalty to -1.
531 * Added queue option autopausedelay when autopause is enabled it will be
532 delayed for this number of seconds since last successful call if there
533 was no prior call the agent will be autopaused immediately.
534 * Added member option ignorebusy this when set and ringinuse is not
535 will allow per member control of multiple calls as ringinuse does for
537 * Added global option check_state_unknown to enforce checking of device state
538 when the device state is unknown app_queue will see unknown as available.
542 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
544 * Added 'k' option to MeetMe to automatically kill the conference when there's only
545 one participant left (much like a normal call bridge)
546 * Added extra argument to Originate to set timeout.
550 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
551 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
552 utility in the UTILS section of menuselect. If an existing astdb is found and no
553 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
554 convert an existing astdb to the SQLite3 version automatically at runtime.
558 * Modules marked as deprecated are no longer marked as building by default. Enabling
559 these modules is still available via menuselect.
563 * authdebug is now disabled by default. To enable this functionaility again
564 set authdebug = yes in iax.conf.
568 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
569 releases it was disabled.
573 * The PBX core previously made a call with a non-existing extension test for
574 extension s@default and jump there if the extension existed.
575 This was a bad default behaviour and violated the principle of least surprise.
576 It has therefore been changed in this release. It may affect some
577 applications and configurations that rely on this behaviour. Most channel
578 drivers have avoided this for many releases by testing whether the extension
579 called exists before starting the PBX and generating a local error.
580 This behaviour still exists and works as before.
582 Extension "s" is used when no extension is given in a channel driver,
583 like immediate answer in DAHDI or calling to a domain with no user part
586 ------------------------------------------------------------------------------
587 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
588 ------------------------------------------------------------------------------
592 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
593 now defaults to force_rport. It is very important that phones requiring nat=no be
594 specifically set as such instead of relying on the default setting. If at all
595 possible, all devices should have nat settings configured in the general section as
596 opposed to configuring nat per-device.
597 * Added preferred_codec_only option in sip.conf. This feature limits the joint
598 codecs sent in response to an INVITE to the single most preferred codec.
599 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
600 to be used for the outgoing call. It must be one of the codecs configured
602 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
603 to be used for holding a private key. If tlsprivatekey is not specified,
604 tlscertfile is searched for both public and private key.
605 * Added tlsclientmethod option to sip.conf. This allows the protocol for
606 outbound client connections to be specified.
607 * The sendrpid parameter has been expanded to include the options
608 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
609 header to be sent (equivalent to setting sendrpid=yes) and setting
610 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
611 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
612 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
613 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
614 will accept the SDP even if the SDP version number is not properly incremented,
615 but will generate a warning in the log indicating that the SIP peer that sent
616 the SDP should have the 'ignoresdpversion' option set.
617 * The 'nat' option has now been been changed to have yes, no, force_rport, and
618 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
619 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
620 remote side requests it and disables symmetric RTP support. Setting it to
621 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
622 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
623 and enables symmetric RTP support.
624 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
625 response. This permits the master channel to know how each channel dialled
626 in a multi-channel setup resolved in an individual way. This carries a
627 performance penalty and can be disabled in sip.conf using the
628 'storesipcause' option.
629 * Added 'externtcpport' and 'externtlsport' options to allow custom port
630 configuration for the externip and externhost options when tcp or tls is used.
631 * Added support for message body (stored in content variable) to SIP NOTIFY message
632 accessible via AMI and CLI.
633 * Added 'media_address' configuration option which can be used to explicitly specify
634 the IP address to use in the SDP for media (audio, video, and text) streams.
635 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
636 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
638 * Added 'use_q850_reason' configuration option for generating and parsing
639 if available Reason: Q.850;cause=<cause code> header. It is implemented
640 in some gateways for better passing PRI/SS7 cause codes via SIP.
641 * When dialing SIP peers, a new component may be added to the end of the dialstring
642 to indicate that a specific remote IP address or host should be used when dialing
643 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
644 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
645 ability to selectively force bridged channels to also be encrypted is also
646 implemented. Branching in the dialplan can be done based on whether or not
647 a channel has secure media and/or signaling.
648 * Added directmediapermit/directmediadeny to limit which peers can send direct media
650 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
651 Charge messages to snom phones.
652 * Added support for G.719 media streams.
653 * Added support for 16khz signed linear media streams.
654 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
655 RTP has been outfitted with the same abilities.
656 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
657 available in device configurations as well as in the dial plan.
658 * Addition of the 'subscribe_network_change' option for turning on and off
659 res_stun_monitor module support in chan_sip.
660 * Addition of the 'auth_options_requests' option for turning on and off
661 authentication for OPTIONS requests in chan_sip.
665 * Add #tryinclude statement for config files. This provides the same
666 functionality as the #include statement however an asterisk module will
667 still load if the filename does not exist. Using the #include statement
668 Asterisk will not allow the module to load.
672 * Added rtsavesysname option into iax.conf to allow the systname to be saved
674 * Added the ability for chan_iax2 to inform the dialplan whether or not
675 encryption is being used. This interoperates with the SIP SRTP implementation
676 so that a secure SIP call can be bridged to a secure IAX call when the
677 dialplan requires bridged channels to be "secure".
678 * Addition of the 'subscribe_network_change' option for turning on and off
679 res_stun_monitor module support in chan_iax.
684 * Added ability to preset channel variables on indicated lines with the setvar
685 configuration option. Also, clearvars=all resets the list of variables back
687 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
688 See configs/res_pktccops.conf for more information.
690 XMPP Google Talk/Jingle changes
691 -------------------------------
692 * Added the externip option to gtalk.conf.
693 * Added the stunaddr option to gtalk.conf which allows for the automatic
694 retrieval of the external ip from a stun server.
698 * Added 'p' option to PickupChan() to allow for picking up channel by the first
699 match to a partial channel name.
700 * Added .m3u support for Mp3Player application.
701 * Added progress option to the app_dial D() option. When progress DTMF is
702 present, those values are sent immediately upon receiving a PROGRESS message
703 regardless if the call has been answered or not.
704 * Added functionality to the app_dial F() option to continue with execution
705 at the current location when no parameters are provided.
706 * Added the 'a' option to app_dial to answer the calling channel before any
707 announcements or macros are executed.
708 * Modified app_dial to set answertime when the called channel answers even if
709 the called channel hangs up during playback of an announcement.
710 * Modified app_dial 'r' option to support an additional parameter to play an
711 indication tone from indications.conf
712 * Added c() option to app_chanspy. This option allows custom DTMF to be set
713 to cycle through the next available channel. By default this is still '*'.
714 * Added x() option to app_chanspy. This option allows DTMF to be set to
715 exit the application.
716 * The Voicemail application has been improved to automatically ignore messages
717 that only contain silence.
718 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
719 associated mailbox(es) to be greetings-only.
720 * The ChanSpy application now has the 'S' option, which makes the application
721 automatically exit once it hits a point where no more channels are available
723 * The ChanSpy application also now has the 'E' option, which spies on a single
724 channel and exits when that channel hangs up.
725 * The MeetMe application now turns on the DENOISE() function by default, for
726 each participant. In our tests, this has significantly decreased background
727 noise (especially noisy data centers).
728 * Voicemail now permits storage of secrets in a separate file, located in the
729 spool directory of each individual user. The control for this is located in
730 the "passwordlocation" option in voicemail.conf. Please see the sample
731 configuration for more information.
732 * The ChanIsAvail application now exposes the returned cause code using a separate
733 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
734 * Added 'd' option to app_followme. This option disables the "Please hold"
736 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
737 received will terminate recording.
738 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
739 Previously the folder could only be set per context, but has now been extended
740 using the imapfolder option.
741 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
742 * Voicemail now allows the pager date format to be specified separately from the
744 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
745 to allow joining, leaving, and sending text to group chats.
746 * MeetMe has a new option 'G' to play an announcement before joining a conference.
747 * Page has a new option 'A(x)' which will playback an announcement simultaneously
748 to all paged phones (and optionally excluding the caller's one using the new
749 option 'n') before the call is bridged.
750 * The 'f' option to Dial has been augmented to take an optional argument. If no
751 argument is provided, the 'f' option works as it always has. If an argument is
752 provided, then the connected party information of all outgoing channels created
753 during the Dial will be set to the argument passed to the 'f' option.
754 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
756 * The OSP lookup application adds in/outbound network ID, optional security,
757 number portability, QoS reporting, destination IP port, custom info and service
759 * Added new application VMSayName that will play the recorded name of the voicemail
760 user if it exists, otherwise will play the mailbox number.
761 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
762 retrieve state for a particular bridge, where <name> is the conference name
763 * app_directory now allows exiting at any time using the operator or pound key.
764 * Voicemail now supports setting a locale per-mailbox.
765 * Two new applications are provided for declining counting phrases in multiple
766 languages. See the application notes for SayCountedNoun and SayCountedAdj for
768 * Voicemail now runs the externnotify script when pollmailboxes is activated and
770 * Voicemail now includes rdnis within msgXXXX.txt file.
771 * ExternalIVR now supports IPv6 addresses.
772 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
773 at https://wiki.asterisk.org/wiki/x/oQBB
774 * ParkedCall and Park can now specify the parking lot to use.
778 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
779 over SRV records associated with a specific service. From the CLI, type
780 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
781 details on how these may be used.
782 * PITCH_SHIFT dialplan function added. This function can be used to modify the
783 pitch of a channel's tx and rx audio streams.
784 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
785 setting various connected line and redirecting party information.
786 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
787 support ISDN subaddressing.
788 * The CHANNEL() function now supports the "name" and "checkhangup" options.
789 * For DAHDI channels, the CHANNEL() dialplan function now allows
790 the dialplan to request changes in the configuration of the active
791 echo canceller on the channel (if any), for the current call only.
794 exten => s,n,Set(CHANNEL(echocan_mode)=off)
796 The possible values are:
798 on - normal mode (the echo canceller is actually reinitialized)
800 fax - FAX/data mode (NLP disabled if possible, otherwise completely
802 voice - voice mode (returns from FAX mode, reverting the changes that
803 were made when FAX mode was requested)
804 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
805 and setting variables on the channel which created the current channel.
806 Administrators should take care to avoid naming conflicts, when multiple
807 channels are dialled at once, especially when used with the Local channel
808 construct (which all could set variables on the master channel). Usage
809 of the HASH() dialplan function, with the key set to the name of the slave
810 channel, is one approach that will avoid conflicts.
811 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
813 * func_odbc now allows multiple row results to be retrieved without using
814 mode=multirow. If rowlimit is set, then additional rows may be retrieved
815 from the same query by using the name of the function which retrieved the
816 first row as an argument to ODBC_FETCH().
817 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
818 dialplan. This function returns the content of the received message.
819 * Added REPLACE, which searches a given variable name for a set of characters,
820 then either replaces them with a single character or deletes them.
821 * Added PASSTHRU, which literally passes the same argument back as its return
822 value. The intent is to be able to use a literal string argument to
823 functions that currently require a variable name as an argument.
824 * HASH-associated variables now can be inherited across channel creation, by
825 prefixing the name of the hash at assignment with the appropriate number of
826 underscores, just like variables.
827 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
828 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
829 whether or not channels that are bridged to the current channel will be
830 required to have secure signaling and/or media.
831 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
832 the current channel has secure signaling and/or media.
833 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
834 "no_media_path" option.
835 Returns "0" if there is a B channel associated with the call.
836 Returns "1" if no B channel is associated with the call. The call is either
837 on hold or is a call waiting call.
838 * Added option to dialplan function CDR(), the 'f' option
839 allows for high resolution times for billsec and duration fields.
840 * FILE() now supports line-mode and writing.
841 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
842 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
846 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
847 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
848 and is set when a dynamic feature is triggered.
849 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
850 to dynamically create a new parking lot matching the value this varible is
852 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
853 features.conf that should be the base for dynamic parkinglots.
854 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
855 parkinglot should have.
856 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
857 parkinglot should have.
858 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
863 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
865 * Added 'R' option to app_queue. This option stops moh and indicates ringing
866 to the caller when an Agent's phone is ringing. This can be used to indicate
867 to the caller that their call is about to be picked up, which is nice when
868 one has been on hold for an extened period of time.
869 * A new config option, penaltymemberslimit, has been added to queues.conf.
870 When set this option will disregard penalty settings when a queue has too
872 * A new option, 'I' has been added to both app_queue and app_dial.
873 By setting this option, Asterisk will not update the caller with
874 connected line changes or redirecting party changes when they occur.
875 * A 'relative-periodic-announce' option has been added to queues.conf. When
876 enabled, this option will cause periodic announce times to be calculated
877 from the end of announcements rather than from the beginning.
878 * The autopause option in queues.conf can be passed a new value, "all." The
879 result is that if a member becomes auto-paused, he will be paused in all
880 queues for which he is a member, not just the queue that failed to reach
882 * Added dialplan function QUEUE_EXISTS to check if a queue exists
883 * The queue logger now allows events to optionally propagate to a file,
884 even when realtime logging is turned on. Additionally, realtime logging
885 supports sending the event arguments to 5 individual fields, although it
886 will fallback to the previous data definition, if the new table layout is
889 mISDN channel driver (chan_misdn) changes
890 ----------------------------------------
891 * Added display_connected parameter to misdn.conf to put a display string
892 in the CONNECT message containing the connected name and/or number if
893 the presentation setting permits it.
894 * Added display_setup parameter to misdn.conf to put a display string
895 in the SETUP message containing the caller name and/or number if the
896 presentation setting permits it.
897 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
898 indicate the dialplan settings are to be obtained from the asterisk
900 * Made misdn.conf parameter callerid accept the "name" <number> format
901 used by the rest of the system.
902 * Made use the nationalprefix and internationalprefix misdn.conf
903 parameters to prefix any received number from the ISDN link if that
904 number has the corresponding Type-Of-Number. NOTE: This includes
905 comparing the incoming call's dialed number against the MSN list.
906 * Added the following new parameters: unknownprefix, netspecificprefix,
907 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
908 received number from the ISDN link if that number has the corresponding
910 * Added new dialplan application misdn_command which permits controlling
911 the CCBS/CCNR functionality.
912 * Added new dialplan function mISDN_CC which permits retrieval of various
913 values from an active call completion record.
914 * For PTP, you should manually send the COLR of the redirected-to party
915 for an incomming redirected call if the incoming call could experience
916 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
917 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
918 if the REDIRECTING(from-num) is not empty.
919 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
920 option on all of the REDIRECTING statements before dialing the
921 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
922 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
923 redirecting-to presentation (COLR) when it becomes available.
924 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
927 thirdparty mISDN enhancements
928 -----------------------------
929 mISDN has been modified by Digium, Inc. to greatly expand facility message
931 * Enhanced COLP support for call diversion and transfer.
934 The latest modified mISDN v1.1.x based version is available at:
935 http://svn.digium.com/svn/thirdparty/mISDN/trunk
936 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
938 Tagged versions of the modified mISDN code are available under:
939 http://svn.digium.com/svn/thirdparty/mISDN/tags
940 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
942 libpri channel driver (chan_dahdi) DAHDI changes
943 -------------------------------------------
944 * The channel variable PRIREDIRECTREASON is now just a status variable
945 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
946 to read and alter the reason.
947 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
948 redirected-to party for an incomming redirected call if the incoming call
949 could experience further redirects. Just set the
950 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
951 to the COLR. A call has been redirected if the REDIRECTING(count) is not
953 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
954 use the inhibit(i) option on all of the REDIRECTING statements before
955 dialing the redirected-to party. You still have to set the
956 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
957 will update the redirecting-to presentation (COLR) when it becomes available.
958 * Added the ability to ignore calls that are not in a Multiple Subscriber
959 Number (MSN) list for PTMP CPE interfaces.
960 * Added dynamic range compression support for dahdi channels. It is
961 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
962 * Added support for ISDN calling and called subaddress with partial support
963 for connected line subaddress.
964 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
965 * Added handling of received HOLD/RETRIEVE messages and the optional ability
966 to transfer a held call on disconnect similar to an analog phone.
967 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
968 Will reroute/deflect an outgoing call when receive the message.
969 Can use the DAHDISendCallreroutingFacility to send the message for the
971 * Added standard location to add options to chan_dahdi dialing:
972 Dial(DAHDI/g1[/extension[/options]])
975 R Reverse charging indication
976 * Added Reverse Charging Indication (Collect calls) send/receive option.
977 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
978 Dial(DAHDI/g1/extension/R)
979 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
980 (requires latest LibPRI)
981 * Added ability to send/receive keypad digits in the SETUP message.
982 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
983 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
984 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
985 (requires latest LibPRI)
986 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
987 to eliminate tromboned calls. A tromboned call goes out an interface and comes
988 back into the same interface. Tromboned calls happen because of call routing,
989 call deflection, call forwarding, and call transfer.
990 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
991 * Added the ability to support call waiting calls. (The SETUP has no B channel
993 * Added Malicious Call ID (MCID) event to the AMI call event class.
994 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
996 Asterisk Manager Interface
997 --------------------------
998 * The Hangup action now accepts a Cause header which may be used to
999 set the channel's hangup cause.
1000 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
1001 to specify a separate .pem file to hold a private key. By default sslcert
1002 is used to hold both the public and private key.
1003 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
1004 for options containing the 'tls' prefix. For example, 'sslenable' is now
1005 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
1006 across all .conf files. All affected sample.conf files have been modified to
1007 reflect this change. Previous options such as 'sslenable' still work,
1008 but options with the 'tls' prefix are preferred.
1009 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
1010 in a channel. (res_mutestream.so)
1011 * The configuration file manager.conf now supports a channelvars option, which
1012 specifies a list of channel variables to include in each channel-oriented
1014 * The redirect command now has new parameters ExtraContext, ExtraExtension,
1015 and ExtraPriority to allow redirecting the second channel to a different
1016 location than the first.
1017 * Added new event "JabberStatus" in the Jabber module to monitor buddies
1019 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
1020 in a MixMonitor recording.
1021 * The 'iax2 show peers' output is now similar to the expected output of
1023 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
1025 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
1026 AOC-E messages on a channel.
1027 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
1028 conform more closely to similar events.
1029 * Added a new eventfilter option per user to allow whitelisting and blacklisting
1031 * Added optional parkinglot variable for park command.
1032 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
1033 if CallerIDNum and CallerIDName headers are also present.
1035 Channel Event Logging
1036 ---------------------
1037 * A new interface, CEL, is introduced here. CEL logs single events, much like
1038 the AMI, but it differs from the AMI in that it logs to db backends much
1039 like CDR does; is based on the event subsystem introduced by Russell, and
1040 can share in all its benefits; allows multiple backends to operate like CDR;
1041 is specialized to event data that would be of concern to billing sytems,
1042 like CDR. Backends for logging and accounting calls have been produced,
1043 but a new CDR backend is still in development.
1047 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
1048 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
1049 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
1050 * Multiple files and formats can now be specified in cdr_custom.conf.
1051 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
1052 See configs/cdr_syslog.conf.sample for more information.
1053 * A 'sequence' field has been added to CDRs which can be combined with
1054 linkedid or uniqueid to uniquely identify a CDR.
1055 * Handling of billsec and duration field has changed. If your table definition
1056 specifies those fields as float,double or similar they will now be logged with
1057 microsecond accuracy instead of a whole integer.
1059 Calendaring for Asterisk
1060 ------------------------
1061 * A new set of modules were added supporing calendar integration with Asterisk.
1062 Dialplan functions for reading from and writing to calendars are included,
1063 as well as the ability to execute dialplan logic upon calendar event notifications.
1064 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
1065 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
1066 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
1067 2003 support does not support forms-based authentication).
1069 Call Completion Supplementary Services for Asterisk
1070 ---------------------------------------------------
1071 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
1072 DAHDI/ISDN supports call completion for the following switch types:
1073 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
1074 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
1076 Multicast RTP Support
1077 ---------------------
1078 * A new RTP engine and channel driver have been added which supports Multicast RTP.
1079 The channel driver can be used with the Page application to perform multicast RTP
1080 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
1081 Type can be either basic or linksys.
1082 Destination is the IP address and port for the RTP packets.
1083 Control address is specific to the linksys type and is used for sending the control
1084 packets unique to them.
1086 Security Events Framework
1087 -------------------------
1088 * Asterisk has a new C API for reporting security events. The module res_security_log
1089 sends these events to the "security" logger level. Currently, AMI is the only
1090 Asterisk component that reports security events. However, SIP support will be
1091 coming soon. For more information on the security events framework, see the
1092 "Asterisk Security Framework" section of the Asterisk wiki at
1093 https://wiki.asterisk.org/wiki/x/wgBQ
1094 * SIP support was added in Asterisk 10
1095 * This API now supports IPv6 addresses
1099 * A technology independent fax frontend (res_fax) has been added to Asterisk.
1100 * A spandsp based fax backend (res_fax_spandsp) has been added.
1101 * The app_fax module has been deprecated in favor of the res_fax module and
1102 the new res_fax_spandsp backend.
1103 * The SendFAX and ReceiveFAX applications now send their log messages to a
1104 'fax' logger level, instead of to the generic logger levels. To see these
1105 messages, the system's logger.conf file will need to direct the 'fax' logger
1106 level to one or more destinations; the logger.conf.sample file includes an
1107 example of how to do this. Note that if the 'fax' logger level is *not*
1108 directed to at least one destination, log messages generated by these
1109 applications will be lost, and that if the 'fax' logger level is directed to
1110 the console, the 'core set verbose' and 'core set debug' CLI commands will
1111 have no effect on whether the messages appear on the console or not.
1115 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
1116 Now, in order to enable transmitting silence during record the transmit_silence
1117 option should be used. transmit_silence_during_record remains a valid option, but
1118 defaults to the behavior of the transmit_silence option.
1119 * Addition of the Unit Test Framework API for managing registration and execution
1120 of unit tests with the purpose of verifying the operation of C functions.
1121 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
1122 XMPP text messages to the remote JID.
1123 * Modules.conf has a new option - "require" - that marks a module as critical for
1124 the execution of Asterisk.
1125 If one of the required modules fail to load, Asterisk will exit with a return
1127 * An 'X' option has been added to the asterisk application which enables #exec support.
1128 This allows #exec to be used in asterisk.conf.
1129 * jabber.conf supports a new option auth_policy that toggles auto user registration.
1130 * A new lockconfdir option has been added to asterisk.conf to protect the
1131 configuration directory (/etc/asterisk by default) during reloads.
1132 * The parkeddynamic option has been added to features.conf to enable the creation
1133 of dynamic parkinglots.
1134 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
1135 the reportalarms config option.
1136 * chan_dahdi supports dialing configuring and dialing by device file name.
1137 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
1138 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
1139 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
1140 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
1141 Handy for the above name-based syntax as it does not depend on
1142 initialization order.
1143 * The Realtime dialplan switch now caches entries for 1 second. This provides a
1144 significant increase in performance (about 3X) for installations using this switchtype.
1145 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
1146 AIS. For more information, please see the Distributed Device State section of the
1147 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1148 * The addition of G.719 pass-through support.
1149 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
1150 during device configuration.
1151 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
1152 have less than 3 lines on the LCD.
1153 * Realtime now supports database failover. See the sample extconfig.conf for details.
1154 * The addition of improved translation path building for wideband codecs. Sample
1155 rate changes during translation are now avoided unless absolutely necessary.
1156 * The addition of the res_stun_monitor module for monitoring and reacting to network
1157 changes while behind a NAT.
1161 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
1162 optionally accept a filename, to apply the setting only to the code generated from
1163 that source file when Asterisk was built. However, there are some modules in Asterisk
1164 that are composed of multiple source files, so this did not result in the behavior
1165 that users expected. In this version, 'core set debug' and 'core set verbose'
1166 can optionally accept *module* names instead (with or without the .so extension),
1167 which applies the setting to the entire module specified, regardless of which source
1168 files it was built from.
1169 * New 'manager show settings' command showing the current settings loaded from
1171 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
1172 the channel hangup request to all channels.
1173 * Added a "core reload" CLI command that executes a global reload of Asterisk.
1175 ------------------------------------------------------------------------------
1176 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
1177 ------------------------------------------------------------------------------
1181 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
1182 Snom phones use this for call pickup of extensions that the phone is
1184 * Added support for setting the domain in the URI for caller of an
1185 outbound call by using the SIPFROMDOMAIN channel variable.
1186 * Added a new configuration option "remotesecret" for authentication to
1187 remote services. For backwards compatibility, "secret" still has the
1188 same function as before, but now you can configure both a remote secret and a
1189 local secret for mutual authentication.
1190 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
1191 the sound will be played to the target of an attended transfer
1192 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
1193 finer control over how many peers Asterisk will qualify and the gap between them
1194 when all peers need to be qualified at the same time.
1195 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
1196 (either globally or for a specific peer), chan_sip will treat any SDP data
1197 it receives as new data and update the media stream accordingly. By
1198 default, Asterisk will only modify the media stream if the SDP session
1199 version received is different from the current SDP session version. This
1200 option is required to interoperate with devices that have non-standard SDP
1201 session version implementations (observed with Microsoft OCS). This option
1202 is disabled by default.
1203 * The parsing of register => lines in sip.conf has been modified to allow a port
1204 to be present in the "user" portion. Please see the sip.conf.sample file for more
1206 * Added support for subscribing to MWI on a remote server and making the status available
1207 as a mailbox. Please see the sip.conf.sample file for more information.
1208 * Added a function to remove SIP headers added in the dialplan before the
1209 first INVITE is generated - SIPRemoveHeader()
1210 * Channel variables set with setvar= in a device configuration is now
1211 set both for inbound and outbound calls.
1212 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
1216 * Added immediate option to iax.conf
1217 * Added forceencryption option to iax.conf
1218 * Added Encryption and Trunk status to manager command "iaxpeers"
1222 * The configuration file now holds separate sections for devices and lines.
1223 Please have a look at configs/skinny.conf.sample and change your skinny.conf
1228 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
1229 support for LibOpenR2. http://www.libopenr2.org/
1230 * The UK option waitfordialtone has been added for use with BT analog
1232 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
1233 is used in conjunction with the 'faxdetect' configuration option. When
1234 'faxbuffers' is used and fax tones are detected, the channel will dynamically
1235 switch to the configured faxbuffers policy. For example, to use 6 buffers
1236 and a 'full' buffer policy for a fax transmission, add:
1238 The faxbuffers configuration will be in affect until the call is torn down.
1239 * Added service message support for 4ESS/5ESS switches.
1243 * For DAHDI channels, the CHANNEL() dialplan function now
1244 supports changing the channel's buffer policy (for the current
1245 call only), using this syntax:
1247 exten => s,n,Set(CHANNEL(buffers)=6,full)
1249 This would change the channel to the 'full' buffer policy and
1250 6 (six) buffers. Possible options for this setting are the same
1251 as those in chan_dahdi.conf.
1252 * Added a new dialplan function, CURLOPT, which permits setting various
1253 options that may be useful with the CURL dialplan function, such as
1254 cookies, proxies, connection timeouts, passwords, etc.
1255 * Permit the syntax and synopsis fields of the corresponding dialplan
1256 functions to be individually set from func_odbc.conf.
1257 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
1258 * func_odbc now may specify an insert query to execute, when the write query
1259 affects 0 rows (usually indicating that no such row exists).
1260 * Added a new dialplan function, LISTFILTER, which permits removing elements
1261 from a set list, by name. Uses the same general syntax as the existing CUT
1262 and FIELDQTY dialplan functions, which also manage lists.
1263 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
1264 obtaining realtime data from the dialplan.
1265 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
1266 a subroutine when using the GoSub() and Return() applications.
1267 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
1268 of "core show function AUDIOHOOK_INHERIT" from the CLI
1269 * Added AES_ENCRYPT. For information on its use, please see the output
1270 of "core show function AES_ENCRYPT" from the CLI
1271 * Added AES_DECRYPT. For information on its use, please see the output
1272 of "core show function AES_DECRYPT" from the CLI
1273 * func_odbc now supports database transactions across multiple queries.
1277 * Scheduled meetme conferences may now have their end times extended by
1279 * app_authenticate now gives the ability to select a prompt other than
1281 * app_directory now pays attention to the searchcontexts setting in
1282 voicemail.conf and will look through all contexts, if no context is
1283 specified in the initial argument.
1284 * A new application, Originate, has been introduced, that allows asynchronous
1285 call origination from the dialplan.
1286 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
1287 in addition to the setting in the "general" context.
1288 * Added ConfBridge dialplan application which does conference bridges without
1289 DAHDI. For information on its use, please see the output of
1290 "core show application ConfBridge" from the CLI.
1294 * The Asterisk CLI has a new command, "channel redirect", which is similar in
1295 operation to the AMI Redirect action.
1296 * extensions.conf now allows you to use keyword "same" to define an extension
1297 without actually specifying an extension. It uses exactly the same pattern
1298 as previously used on the last "exten" line. For example:
1299 exten => 123,1,NoOp(something)
1300 same => n,SomethingElse()
1301 * musiconhold.conf classes of type 'files' can now use relative directory paths,
1302 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
1303 * All deprecated CLI commands are removed from the sourcecode. They are now handled
1304 by the new clialiases module. See cli_aliases.conf.sample file.
1305 * Times within timespecs are now accurate down to the minute. This is a change
1306 from historical Asterisk, which only provided timespecs rounded to the nearest
1307 even (read: evenly divisible by 2) minute mark.
1308 * The realtime switch now supports an option flag, 'p', which disables searches for
1310 * In addition to a time range and date range, timespecs now accept a 5th optional
1311 argument, timezone. This allows you to perform time checks on alternate
1312 timezones, especially if those daylight savings time ranges vary from your
1313 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
1315 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
1316 give you the correct output for an asterisk box behind nat. It will give you the
1317 externhost and localnet settings.
1318 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
1319 can connect calls in passthrough mode, as well as record and play back files.
1320 * Successful and unsuccessful call pickup can now be alerted through sounds, by
1321 using pickupsound and pickupfailsound in features.conf.
1322 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
1323 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
1324 instead of the /var/run/asterisk.pid where it used to be. This will make
1325 installs as non-root easier to manage.
1330 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
1331 be written; they will no longer be explicitly written.
1333 Asterisk Manager Interface
1334 --------------------------
1335 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
1336 a non-empty value) in your request. If you do this, any pending AMI events will
1337 *not* be included in the response to your request as they would normally, but
1338 will be left in the event queue for the next request you make to retrieve. For
1339 some applications, this will allow you to guarantee that you will only see
1340 events in responses to 'WaitEvent' actions, and can better know when to expect them.
1341 To know whether the Asterisk server supports this header or not, your client can
1342 inspect the first response back from the server to see if it includes this header:
1344 Pragma: SuppressEvents
1346 If this is included, the server supports event suppression.
1348 * Added 4 new Actions to list skinny device(s) and line(s)
1354 LDAP Schema File Additions
1355 --------------------------
1356 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
1357 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
1359 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
1360 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
1361 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
1362 * Removed redundant IPaddr (there's already IPAddress)
1363 - Gives more configuration Flags for SIP-Users available (tested)
1364 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
1365 without extensibleObject (which really should be the last resort); gives
1366 also additional possibilities for LDAP-filter
1368 ------------------------------------------------------------------------------
1369 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
1370 ------------------------------------------------------------------------------
1372 Device State Handling
1373 ---------------------
1374 * The event infrastructure in Asterisk got another big update to help support
1375 distributed events. It currently supports distributed device state and
1376 distributed Voicemail MWI (Message Waiting Indication). A new module has
1377 been merged, res_ais, which facilitates communicating events between servers.
1378 It uses the SAForum AIS (Service Availability Forum Application Interface
1379 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
1380 a cluster of Asterisk servers, and to share events between them. For more
1381 information on setting this up, refer to the Distributed Device State section
1382 of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1386 * Added a new dialplan function, AST_CONFIG(), which allows you to access
1387 variables from an Asterisk configuration file.
1388 * The JACK_HOOK function now has a c() option to supply a custom client name.
1389 * Added two new dialplan functions from libspeex for audio gain control and
1390 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
1391 rx directions of a channel from the dialplan.
1392 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
1393 based on other parameters. The default is still to search based on the
1394 forwarding station ID. However, there are new options that allow you to search
1395 based on the message desk terminal ID, or the message desk number.
1396 * TIMEOUT() has been modified to be accurate down to the millisecond.
1397 * ENUM*() functions now include the following new options:
1398 - 'u' returns the full URI and does not strip off the URI-scheme.
1399 - 's' triggers ISN specific rewriting
1400 - 'i' looks for branches into an Infrastructure ENUM tree
1401 - 'd' for a direct DNS lookup without any flipping of digits.
1402 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
1403 * CHANNEL() now has options for the maximum, minimum, and standard or normal
1404 deviation of jitter, rtt, and loss for a call using chan_sip.
1406 DAHDI channel driver (chan_dahdi) Changes
1407 ----------------------------------------
1408 * Channels can now be configured using named sections in chan_dahdi.conf, just
1409 like other channel drivers, including the use of templates.
1410 * The default for pridialplan has changed from 'national' to 'unknown'.
1414 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
1415 to something that matches the pattern a hint will be created using the contents
1416 and variables evaluated.
1417 * Dialplan matching has been extended to allow an extension to return to the
1418 PBX core to wait for more digits. This is done by using the new dialplan
1419 application called "Incomplete". This will permit a whole new level of
1420 extension control, by giving the administrator more control over early
1421 matches employing one of the short-circuit pattern match operators. Note
1422 that custom applications can trigger this same behavior by returning the
1423 special value AST_PBX_INCOMPLETE.
1427 * Directory now permits both first and last names to be matched at the same
1428 time. In addition, the number of digits to enter of the name can be set in
1429 the arguments to Directory; previously, you could enter only 3, regardless
1430 of how many names are in your company. For large companies, this should be
1432 * Voicemail now permits a mailbox setting to wrap around from first to last
1433 messages, if the "messagewrap" option is set to a true value.
1434 * Voicemail now permits an external script to be run, for password validation.
1435 The script should output "VALID" or "INVALID" on stdout, depending upon the
1436 wish to validate or invalidate the password given. Arguments are:
1437 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
1439 * Dial has a new option: F(context^extension^pri), which permits a callee to
1440 continue in the dialplan, at the specified label, if the caller hangs up.
1441 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
1442 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
1443 * The Jack application now has a c() option to supply a custom client name.
1444 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
1445 like the pre-existing whisper mode, except that the spy can also talk to the
1446 participant on the bridged channel as well.
1447 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
1448 to be spoken instead of the channel name or number. For more information on the
1449 use of this option, issue the command "core show application ChanSpy" from the
1451 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
1452 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
1453 words, if using the 'd' option, it is not possible to enter a number to append to
1454 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
1455 change to whisper mode, and pressing 6 will change to barge mode.
1456 * ExternalIVR now takes several options that affect the way it performs, as
1457 well as having several new commands. Please see the External IVR page on the Asterisk
1458 wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
1459 * Added ability to communicate over a TCP socket instead of forking a child process for the
1460 ExternalIVR application.
1461 * ChanIsAvail has a new option, 'a', which will return all available channels instead
1462 of just the first one if you give the function more then one channel to check.
1463 * PrivacyManager now takes an option where you can specify a context where the
1464 given number will be matched. This way you have more control over who is allowed
1465 and it stops the people who blindly enter 10 digits.
1466 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
1467 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
1468 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
1469 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
1470 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
1471 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
1472 * The Dial() application no longer copies the language used by the caller to the callee's
1473 channel. If you desire for the caller's channel's language to be used for file playback
1474 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
1475 * SendImage() no longer hangs up the channel on error; instead, it sets the
1476 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
1477 'UNSUPPORTED'. This change makes SendImage() more consistent with other
1479 * Park has a new option, 's', which silences the announcement of the parking space number.
1480 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
1481 invalid input and will be assumed to mean that no timeout is desired.
1485 * Added DNS manager support to registrations for peers referencing peer entries.
1486 DNS manager runs in the background which allows DNS lookups to be run asynchronously
1487 as well as periodically updating the IP address. These properties allow for
1488 better performance as well as recovery in the event of an IP change.
1489 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
1490 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
1491 These changes also provide performance improvements for call setup and tear down.
1492 * Added ability to specify registration expiry time on a per registration basis in
1494 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
1496 * Added t38pt_usertpsource option. See sip.conf.sample for details.
1497 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
1498 * 'sip show peers' and 'sip show users' display their entries sorted in
1499 alphabetical order, as opposed to the order they were in, in the config
1501 * Videosupport now supports an additional option, "always", which always sets
1502 up video RTP ports, even on clients that don't support it. This helps with
1503 callfiles and certain transfers to ensure that if two video phones are
1504 connected, they will always share video feeds.
1508 * Existing DNS manager lookups extended to check for SRV records.
1509 * IAX2 encryption support has been improved to support periodic key rotation
1510 within a call for enhanced security. The option "keyrotate" has been
1511 provided to disable this functionality to preserve backwards compatibility
1512 with older versions of IAX2 that do not support key rotation.
1516 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
1517 data tree based on the given <path>.
1518 * New CLI command "data show providers" that will display all the registered
1520 * New CLI command, "config reload <file.conf>" which reloads any module that
1521 references that particular configuration file. Also added "config list"
1522 which shows which configuration files are in use.
1523 * New CLI commands, "pri show version" and "ss7 show version" that will
1524 display which version of libpri and libss7 are being used, respectively.
1525 A new API call was added so trunk will now have to be compiled against
1526 a versions of libpri and libss7 that have them or it will not know that
1527 these libraries exist.
1528 * The commands "core show globals", "core set global" and "core set chanvar" has
1529 been deprecated in favor of the more semanticly correct "dialplan show globals",
1530 "dialplan set chanvar" and "dialplan set global".
1531 * New CLI command "dialplan show chanvar" to list all variables associated
1532 with a given channel.
1536 * Addresses managed by DNS manager now can check to see if there is a DNS
1537 SRV record for a given domain and will use that hostname/port if present.
1539 AMI - The manager (TCP/TLS/HTTP)
1540 --------------------------------
1541 * The Status command now takes an optional list of variables to display
1542 along with channel status.
1543 * The QueueEntry event now also includes the channel's uniqueid
1547 * res_odbc no longer has a limit of 1023 total possible unshared connections,
1548 as some people were running into this limit. This limit has been increased
1553 * The TRANSFER queue log entry now includes the the caller's original
1554 position in the transferred-from queue.
1555 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
1556 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
1557 as well as an explanation about timeout options in general
1558 * Added a new option - C - for forcing the "answered elsewhere" flag on
1559 cancellation of calls in to members of the queue. This is to avoid the
1560 call to a member of a queue having the call listed as a "missed call".
1564 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
1565 adaptive capabilities. What this means in practical terms is that if your
1566 realtime table lacks critical fields, Asterisk will now emit warnings to
1567 that effect. Also, some of the realtime drivers have the ability (if
1568 configured) to automatically add those columns to the table with the
1569 correct type and length.
1573 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
1574 the 'setvar' option to cause a given audio file to be played upon completion
1575 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
1576 Skinny channels only.
1577 * You can now compile Asterisk against the Hoard Memory Allocator, see the
1578 Hoard page on the Asterisk wiki for more information:
1579 https://wiki.asterisk.org/wiki/x/pQBB
1580 * Config file variables may now be appended to, by using the '+=' append
1581 operator. This is most helpful when working with long SQL queries in
1582 func_odbc.conf, as the queries no longer need to be specified on a single
1584 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
1585 which will add a second to the billsec when the ending
1586 time is set, if the number in the microseconds field of the end time is
1587 greater than the number of microseconds in the answer time. This allows
1588 users to count the 'initiated' seconds in their billing records.
1590 ------------------------------------------------------------------------------
1591 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
1592 ------------------------------------------------------------------------------
1594 AMI - The manager (TCP/TLS/HTTP)
1595 --------------------------------
1596 * Manager has undergone a lot of changes, all of them documented
1597 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
1598 * Manager version has changed to 1.1
1599 * Added a new action 'CoreShowChannels' to list currently defined channels
1600 and some information about them.
1601 * Added a new action 'SIPshowregistry' to list SIP registrations.
1602 * Added TLS support for the manager interface and HTTP server
1603 * Added the URI redirect option for the built-in HTTP server
1604 * The output of CallerID in Manager events is now more consistent.
1605 CallerIDNum is used for number and CallerIDName for name.
1606 * Enable https support for builtin web server.
1607 See configs/http.conf.sample for details.
1608 * Added a new action, GetConfigJSON, which can return the contents of an
1609 Asterisk configuration file in JSON format. This is intended to help
1610 improve the performance of AJAX applications using the manager interface
1612 * SIP and IAX manager events now use "ChannelType" in all cases where we
1613 indicate channel driver. Previously, we used a mixture of "Channel"
1614 and "ChannelDriver" headers.
1615 * Added a "Bridge" action which allows you to bridge any two channels that
1616 are currently active on the system.
1617 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
1618 the voicemail users setup.
1619 * Added 'DBDel' and 'DBDelTree' manager commands.
1620 * cdr_manager now reports events via the "cdr" level, separating it from
1621 the very verbose "call" level.
1622 * Manager users are now stored in memory. If you change the manager account
1623 list (delete or add accounts) you need to reload manager.
1624 * Added Masquerade manager event for when a masquerade happens between
1626 * Added "manager reload" command for the CLI
1627 * Lots of commands that only provided information are now allowed under the
1628 Reporting privilege, instead of only under Call or System.
1629 * The IAX* commands now require either System or Reporting privilege, to
1630 mirror the privileges of the SIP* commands.
1631 * Added ability to retrieve list of categories in a config file.
1632 * Added ability to retrieve the content of a particular category.
1633 * Added ability to empty a context.
1634 * Created new action to create a new file.
1635 * Updated delete action to allow deletion by line number with respect to category.
1636 * Added new action insert to add new variable to category at specified line.
1637 * Updated action newcat to allow new category to be inserted in file above another
1639 * Added new event "JitterBufStats" in the IAX2 channel
1640 * Originate now requires the Originate privilege and, if you want to call out
1641 to a subshell, it requires the System privilege, as well. This was done to
1642 enhance manager security.
1643 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
1644 * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
1645 or manager show command Atxfer from the CLI
1646 * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
1647 details or manager show command IAXregistry from the CLI
1651 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
1652 state in the dialplan, as well as creating custom device states that are
1653 controllable from the dialplan.
1654 * Extend CALLERID() function with "pres" and "ton" parameters to
1655 fetch string representation of calling number presentation indicator
1656 and numeric representation of type of calling number value.
1657 * MailboxExists converted to dialplan function
1658 * A new option to Dial() for telling IP phones not to count the call
1659 as "missed" when dial times out and cancels.
1660 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
1661 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
1662 held for any given channel. Also, locks are automatically freed when a
1664 * Added HINT() dialplan function that allows retrieving hint information.
1665 Hints are mappings between extensions and devices for the sake of
1666 determining the state of an extension. This function can retrieve the list
1667 of devices or the name associated with a hint.
1668 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
1670 * Added SYSINFO() dialplan function which allows retrieval of system information
1671 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
1672 the existence of a dialplan target.
1673 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
1674 upper and lower case, respectively.
1675 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
1676 ID for the call (not the Asterisk call ID or unique ID), provided that the
1677 channel driver supports this. For SIP, you get the SIP call-ID for the
1678 bridged channel which you can store in the CDR with a custom field.
1682 * Added CLI permissions, config file: cli_permissions.conf
1683 default is to allow all commands for every local user/group.
1684 Also this new feature added three new CLI commands:
1685 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
1686 - cli reload permissions
1687 - cli show permissions
1688 * New CLI command "core show hint" (usage: core show hint <exten>)
1689 * New CLI command "core show settings"
1690 * Added 'core show channels count' CLI command.
1691 * Added the ability to set the core debug and verbose values on a per-file basis.
1692 * Added 'queue pause member' and 'queue unpause member' CLI commands
1693 * Ability to set process limits ("ulimit") without restarting Asterisk
1694 * Enhanced "agi debug" to print the channel name as a prefix to the debug
1695 output to make debugging on busy systems much easier.
1696 * New CLI commands "dialplan set extenpatternmatching true/false"
1697 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
1698 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
1699 listed in the startup_commands section of cli.conf will get executed.
1700 * Added a CLI command, "devstate change", which allows you to set custom device
1701 states from the func_devstate module that provides the DEVICE_STATE() function
1702 and handling of the "Custom:" devices.
1703 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
1704 sorted into the different possible callbacks, with the number of entries
1705 currently scheduled for each. Gives you a feel for how busy the sip channel
1707 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
1708 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
1709 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
1713 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
1714 option is enabled, Asterisk will watch for a CNG tone in the incoming audio
1715 for a received call. If it is detected, the channel will jump to the
1716 'fax' extension in the dialplan.
1717 * The default SIP useragent= identifier now includes the Asterisk version
1718 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
1719 If set, and the incoming request carries authentication info,
1720 the username to match in the users list is taken from the Digest header
1721 rather than from the From: field. This feature is considered experimental.
1722 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
1723 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
1724 * The "localmask" setting was removed in version 1.2 and the reminder about it
1725 being removed is now also removed.
1726 * A new option "busylevel" for setting a level of calls where asterisk reports
1727 a device as busy, to separate it from call-limit. This value is also added
1728 to the SIP_PEER dialplan function.
1729 * A new realtime family called "sipregs" is now supported to store SIP registration
1730 data. If this family is defined, "sippeers" will be used for configuration and
1731 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
1732 registration data, as before.
1733 * The SIPPEER function have new options for port address, call and pickup groups
1734 * Added support for T.140 realtime text in SIP/RTP
1735 * The "checkmwi" option has been removed from sip.conf, as it is no longer
1736 required due to the restructuring of how MWI is handled. See the descriptions
1737 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
1738 for more information.
1739 * Added rtpdest option to CHANNEL() dialplan function.
1740 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
1741 * SIP now adds a header to the CANCEL if the call was answered by another phone
1742 in the same dial command, or if the new c option in dial() is used.
1743 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
1744 states it is not needed. For phones, however, that do require it the "registertrying" option
1745 has been added so it can be enabled.
1746 * A new option called "callcounter" (global/peer/user level) enables call counters needed
1747 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
1748 used to enable this functionality).
1749 * New settings for timer T1 and timer B on a global level or per device. This makes it
1750 possible to force timeout faster on non-responsive SIP servers. These settings are
1751 considered advanced, so don't use them unless you have a problem.
1752 * Added a dial string option to be able to set the To: header in an INVITE to any
1754 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
1755 the qualify frequency.
1756 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
1757 were not properly torn down due to network or endpoint failures during an established
1759 * Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB
1760 and configs/sip.conf.sample for more information on how it is used.
1761 * Added a new configuration option "authfailureevents" that enables manager events when
1762 a peer can't authenticate properly.
1763 * Added DNS manager support to registrations for peers not referencing a peer entry.
1767 * Added the trunkmaxsize configuration option to chan_iax2.
1768 * Added the srvlookup option to iax.conf
1769 * Added support for OSP. The token is set and retrieved through the CHANNEL()
1772 XMPP Google Talk/Jingle changes
1773 -------------------------------
1774 * Added the bindaddr option to gtalk.conf.
1778 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
1779 * Proper codec support in chan_skinny.
1780 * Added settings for IP and Ethernet QoS requests
1784 * Added separate settings for media QoS in mgcp.conf
1786 Console Channel Driver changes
1787 ------------------------------
1788 * Added experimental support for video send & receive to chan_oss.
1789 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
1792 Phone channel changes (chan_phone)
1793 ----------------------------------
1794 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
1796 H.323 channel Changes
1797 ---------------------
1798 * H323 remote hold notification support added (by NOTIFY message
1799 and/or H.450 supplementary service)
1801 Local channel changes
1802 ---------------------
1803 * The device state functionality in the Local channel driver has been updated
1804 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
1805 to just UNKNOWN if the extension exists.
1806 * Added jitterbuffer support for chan_local. This allows you to use the
1807 generic jitterbuffer on incoming calls going to Asterisk applications.
1808 For example, this would allow you to use a jitterbuffer for an incoming
1809 SIP call to Voicemail by putting a Local channel in the middle. This
1810 feature is enabled by using the 'j' option in the Dial string to the Local
1811 channel in conjunction with the existing 'n' option for local channels.
1812 * A 'b' option has been added which causes chan_local to return the actual channel
1813 that is behind it when queried. This is useful for transfer scenarios as the
1814 actual channel will be transferred, not the Local channel.
1816 Agent channel changes
1817 ----------------------
1818 * The ackcall and endcall options are now supplemented with options acceptdtmf
1819 and enddtmf. These allow for the DTMF keypress to be configurable. The options
1820 default to their old hard-coded values ('#' and '*' respectively) so this should
1821 not break any existing agent installations.
1823 DAHDI channel driver (chan_dahdi) Changes
1824 ----------------------------------------
1825 * SS7 support (via libss7 library)
1826 * In India, some carriers transmit CID via dtmf. Some code has been added
1827 that will handle some situations. The cidstart=polarity_IN choice has been added for
1828 those carriers that transmit CID via dtmf after a polarity change.
1829 * CID matching information is now shown when doing 'dialplan show'.
1830 * Added dahdi show version CLI command.
1831 * Added setvar support to chan_dahdi.conf channel entries.
1832 * Added two new options: mwimonitor and mwimonitornotify. These options allow
1833 you to enable MWI monitoring on FXO lines. When the MWI state changes,
1834 the script specified in the mwimonitornotify option is executed. An internal
1835 event indicating the new state of the mailbox is also generated, so that
1836 the normal MWI facilities in Asterisk work as usual.
1837 * Added signalling type 'auto', which attempts to use the same signalling type
1838 for a channel as configured in DAHDI. This is primarily designed for analog
1839 ports, but will also work for digital ports that are configured for FXS or FXO
1840 signalling types. This mode is also the default now, so if your chan_dahdi.conf
1841 does not specify signalling for a channel (which is unlikely as the sample
1842 configuration file has always recommended specifying it for every channel) then
1843 the 'auto' mode will be used for that channel if possible.
1844 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
1845 state for a channel; also ensured that the DNDState Manager event is
1846 emitted no matter how the DND state is set or cleared.
1850 * Added a new channel driver, chan_unistim. See the Asterisk wiki at
1851 https://wiki.asterisk.org/wiki/x/vgsiAQ and configs/unistim.conf.sample
1852 for details. This new channel driver allows you to use Nortel i2002,
1853 i2004, and i2050 phones with Asterisk.
1854 * Added a new channel driver, chan_console, which uses portaudio as a cross
1855 platform audio interface. It was written as a channel driver that would
1856 work with Mac CoreAudio, but portaudio supports a number of other audio
1857 interfaces, as well. Note that this channel driver requires v19 or higher
1858 of portaudio; older versions have a different API.
1862 * Added the ability to specify arguments to the Dial application when using
1863 the DUNDi switch in the dialplan.
1864 * Added the ability to set weights for responses dynamically. This can be
1865 done using a global variable or a dialplan function. Using the SHELL()
1866 function would allow you to have an external script set the weight for
1868 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
1869 functions will allow you to initiate a DUNDi query from the dialplan,
1870 find out how many results there are, and access each one.
1871 * Added the ability to specifiy a port for a dundi peer.
1875 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
1876 functions will allow you to initiate an ENUM lookup from the dialplan,
1877 and Asterisk will cache the results. ENUMRESULT can be used to access
1878 the results without doing multiple DNS queries.
1882 * Added the ability to customize which sound files are used for some of the
1883 prompts within the Voicemail application by changing them in voicemail.conf
1884 * Added the ability for the "voicemail show users" CLI command to show users
1885 configured by the dynamic realtime configuration method.
1886 * MWI (Message Waiting Indication) handling has been significantly
1887 restructured internally to Asterisk. It is now totally event based
1888 instead of polling based. The voicemail application will notify other
1889 modules that have subscribed to MWI events when something in the mailbox
1891 This also means that if any other entity outside of Asterisk is changing
1892 the contents of mailboxes, then the voicemail application still needs to
1893 poll for changes. Examples of situations that would require this option
1894 are web interfaces to voicemail or an email client in the case of using
1895 IMAP storage. So, two new options have been added to voicemail.conf
1896 to account for this: "pollmailboxes" and "pollfreq". See the sample
1897 configuration file for details.
1898 * Added "tw" language support
1899 * Added support for storage of greetings using an IMAP server
1900 * Added ability to customize forward, reverse, stop, and pause keys for message playback
1901 * SMDI is now enabled in voicemail using the smdienable option.
1902 * A "lockmode" option has been added to asterisk.conf to configure the file
1903 locking method used for voicemail, and potentially other things in the
1904 future. The default is the old behavior, lockfile. However, there is a
1905 new method, "flock", that uses a different method for situations where the
1906 lockfile will not work, such as on SMB/CIFS mounts.
1907 * Added the ability to backup deleted messages, to ease recovery in the case
1908 that a user accidentally deletes a message, and discovers that they need it.
1909 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
1910 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
1911 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
1912 voicemail boxes. The SMDI interface can also poll for MWI changes when some
1913 outside entity is modifying the state of the mailbox (such as IMAP storage or
1914 a web interface of some kind).
1915 * Added the support for marking messages as "urgent." There are two methods to accomplish
1916 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
1917 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
1918 the message as urgent after he has recorded a voicemail by following the voice instructions.
1919 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
1924 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
1925 used across multiple queues.
1926 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
1927 setqueueentryvar options for each queue, see queues.conf.sample for details.
1928 * Added keepstats option to queues.conf which will keep queue
1929 statistics during a reload.
1930 * setinterfacevar option in queues.conf also now sets a variable
1931 called MEMBERNAME which contains the member's name.
1932 * Added 'Strategy' field to manager event QueueParams which represents
1933 the queue strategy in use.
1934 * Added option to run macro when a queue member is connected to a caller,
1935 see queues.conf.sample for details.
1936 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
1937 does not count paused queue members as unavailable.
1938 * Added min-announce-frequency option to queues.conf which allows you to control the
1939 minimum amount of time between queue announcements for use when the caller's queue
1940 position changes frequently.
1941 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
1943 * Added ability for non-realtime queues to have realtime members
1944 * Added the "linear" strategy to queues.
1945 * Added the "wrandom" strategy to queues.
1946 * Added new channel variable QUEUE_MIN_PENALTY
1947 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
1948 rules in queuerules.conf. See configs/queuerules.conf.sample for details
1949 * Added a new parameter for member definition, called state_interface. This may be
1950 used so that a member may be called via one interface but have a different interface's
1951 device state reported.
1952 * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
1953 "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
1954 "manager show command QueueReset."
1955 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
1956 specified by the periodic-announce option, then one will be chosen randomly when it is time
1957 to play a periodic announcment
1958 * New configuration options: announce-position now takes two more values in addition to "yes" and
1959 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
1960 announce-position-limit. By setting announce-position to "limit" callers will only have their
1961 position announced if their position is less than what is specified by announce-position-limit.
1962 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
1963 will be told that their are more than announce-position-limit callers waiting.
1964 * Two new queue log events have been added. An ADDMEMBER event will be logged
1965 when a realtime queue member is added and a REMOVEMEMBER event will be logged
1966 when a realtime queue member is removed. Since there is no calling channel associated
1967 with these events, the string "REALTIME" is placed where the channel's unique id
1968 is typically placed.
1969 * The configuration method for the "joinempty" and "leavewhenempty" options has
1970 changed to a comma-separated list of methods of determining member availability
1971 instead of vague terms such as "yes," "loose," "no," and "strict." These old four
1972 values are still accepted for backwards-compatibility, though.
1973 * The average talktime is now calculated on queues. This information is reported via the
1974 CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
1975 and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
1980 * The 'o' option to provide an optimization has been removed and its functionality
1981 has been enabled by default.
1982 * When a conference is created, the UNIQUEID of the channel that caused it to be
1983 created is stored. Then, every channel that joins the conference will have the
1984 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
1985 callers that come and go from long standing conferences.
1986 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
1987 except it does operations on a channel by name, instead of number in a conference.
1988 This is a very useful feature in combination with the 'X' option to ChanSpy.
1989 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
1991 * Added new RealTime functionality to provide support for scheduled conferencing.
1992 This includes optional messages to the caller if they attempt to join before
1993 the schedule start time, or to allow the caller to join the conference early.
1994 Also included is optional support for limiting the number of callers per
1995 RealTime conference.
1996 * Added the S() and L() options to the MeetMe application. These are pretty
1997 much identical to the S() and L() options to Dial(). They let you set
1998 timeouts for the conference, as well as have warning sounds played to
1999 let the caller know how much time is left, and when it is running out.
2000 * Added the ability to do "meetme concise" with the "meetme" CLI command.
2001 This extends the concise capabilities of this CLI command to include
2002 listing all conferences, instead of an addition to the other sub commands
2003 for the "meetme" command.
2004 * Added the ability to specify the music on hold class used to play into the
2005 conference when there is only one member and the M option is used.
2006 * Added MEETME_INFO dialplan function which provides a way to query
2007 various properties of a Meetme conference.
2008 * Added new admin features: *81: Roll call, *82: eject all, *83: mute all,
2009 and *84: record in-conf
2011 Other Dialplan Application Changes
2012 ----------------------------------
2013 * Argument support for Gosub application
2014 * From the to-do lists: straighten out the app timeout args:
2015 Wait() app now really does 0.3 seconds- was truncating arg to an int.
2016 WaitExten() same as Wait().
2017 Congestion() - Now takes floating pt. argument.
2018 Busy() - now takes floating pt. argument.
2019 Read() - timeout now can be floating pt.
2020 WaitForRing() now takes floating pt timeout arg.
2021 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
2022 * Added 's' option to Page application.
2023 * Added an optional timeout argument to the Page application.
2024 * Added 'E', 'V', and 'P' commands to ExternalIVR.
2025 * Added 'o' and 'X' options to Chanspy.
2026 * Added a new dialplan application, Bridge, which allows you to bridge the
2027 calling channel to any other active channel on the system.
2028 * Added the ability to specify a music on hold class to play instead of ringing
2029 for the SLATrunk application.
2030 * The Read application no longer exits the dialplan on error. Instead, it sets
2031 READSTATUS to ERROR, which you can catch and handle separately.
2032 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
2033 of asking for verification of each name, one at a time.
2034 * Privacy() no longer uses privacy.conf, as all options are specifyable as
2035 direct options to the app.
2036 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
2038 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
2039 * The ChannelRedirect application no longer exits the dialplan if the given channel
2040 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
2041 or NOCHANNEL if the given channel was not found.
2042 * The silencethreshold setting that was previously configurable in multiple
2043 applications is now settable globally via dsp.conf.
2045 Music On Hold Changes
2046 ---------------------
2047 * A new option, "digit", has been added for music on hold classes in
2048 musiconhold.conf. If this is set for a music on hold class, a caller
2049 listening to music on hold can press this digit to switch to listening
2050 to this music on hold class.
2051 * Support for realtime music on hold has been added.
2052 * In conjunction with the realtime music on hold, a general section has
2053 been added to musiconhold.conf, its sole variable is cachertclasses. If this
2054 is set, then music on hold classes found in realtime will be cached in memory.
2058 * AEL upgraded to use the Gosub with Arguments instead
2059 of Macro application, to hopefully reduce the problems
2060 seen with the artificially low stack ceiling that
2061 Macro bumps into. Macros can only call other Macros
2062 to a depth of 7. Tests run using gosub, show depths
2063 limited only by virtual memory. A small test demonstrated
2064 recursive call depths of 100,000 without problems.
2065 -- in addition to this, all apps that allowed a macro
2066 to be called, as in Dial, queues, etc, are now allowing
2067 a gosub call in similar fashion.
2068 * AEL now generates LOCAL(argname) declarations when it
2069 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
2070 etc. That makes the arguments local in scope. The user
2071 can define their own local variables in macros, now,
2072 by saying "local myvar=someval;" or using Set() in this
2073 fashion: Set(LOCAL(myvar)=someval); ("local" is now
2075 * utils/conf2ael introduced. Will convert an extensions.conf
2076 file into extensions.ael. Very crude and unfinished, but
2077 will be improved as time goes by. Should be useful for a
2078 first pass at conversion.
2079 * aelparse will now read extensions.conf to see if a referenced
2080 macro or context is there before issueing a warning.
2081 * AEL parser sets a local channel variable ~~EXTEN~~, to
2082 preserve the value of ${EXTEN} thru switch statements.
2083 * New operator in $[...] expressions: the ~~ operator serves
2084 as a concatenation operator. AT THE MOMENT, it is really only
2085 necessary and useful in AEL, especially in if() expressions.
2086 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
2087 any enclosing double-quotes, and evaluate to the value of a
2088 concatenated with the value of b. For example if a is set to
2089 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
2090 evaluate to xyzabc .
2093 Call Features (res_features) Changes
2094 ------------------------------------
2095 * Added the parkedcalltransfers option to features.conf
2096 * Added parkedcallparking option to control one touch parking w/ parking
2098 * Added parkedcallhangup option to control disconnect feature w/ parking
2100 * Added parkedcallrecording option to control one-touch record w/ parking
2102 * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
2103 parkedcalltransfers option support for multiple parking lots.
2104 * Added BRIDGE_FEATURES variable to set available features for a channel
2105 * The built-in method for doing attended transfers has been updated to
2106 include some new options that allow you to have the transferee sent
2107 back to the person that did the transfer if the transfer is not successful.
2108 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
2109 in features.conf.sample.
2110 * Added support for configuring named groups of custom call features in
2111 features.conf. This means that features can be written a single time, and
2112 then mapped into groups of features for different key mappings or easier
2114 * Updated the ParkedCall application to allow you to not specify a parking
2115 extension. If you don't specify a parking space to pick up, it will grab
2116 the first one available.
2117 * Added cli command 'features reload' to reload call features from features.conf
2118 * Moved into core asterisk binary.
2119 * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
2120 * Added the ability for custom parking lots to be configured with their own
2121 parking extension with the parkext option.
2123 Language Support Changes
2124 ------------------------
2125 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
2126 * Added support for the Hungarian language for saying numbers, dates, and times.
2130 * Added SPEECH commands for speech recognition. A complete listing can be found
2132 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
2133 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
2134 does not behave as expected; the native command needs to be used, instead.
2135 * Added the ability to perform SRV lookups on fast AGI calls. To use this
2136 feature, simply use hagi: instead of agi: as the protocol portion
2137 of the URI parameter to the AGI function call in your dial plan. Also note
2138 that specifying a port number in the AGI URI will disable SRV lookups,
2139 even if you use the hagi: protocol.
2140 * No longer support MSG_OOB flag on HANGUP.
2144 * Added rotatestrategy option to logger.conf, along with two new options:
2145 "timestamp" which will use the time to name the logger files instead of
2146 sequence number; and "rotate", which rotates the names of the log files,
2147 similar to the way syslog rotates files.
2148 * Added exec_after_rotate option to logger.conf, which allows a system
2149 command to be run after rotation. This is primarily useful with
2150 rotatestrategy=rotate, to allow a limit on the number of log files kept
2151 and to ensure that the oldest log file gets deleted.
2152 * Added realtime support for the queue log
2156 * The cdr_manager module has a [mappings] feature, like cdr_custom,
2157 to add fields to the manager event from the CDR variables.
2158 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
2159 backend database CDR table. Specifically, additional, non-standard
2160 columns are supported, merely by setting the corresponding CDR variable in
2161 your dialplan. In addition, you may alias any column to another name (for
2162 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
2163 simply "alias src => ANI" in the configuration file). Records may be
2164 posted to more than one backend, simply by specifying multiple categories
2165 in the configuration file. And finally, you may filter which CDRs get
2166 posted to each backend, by specifying a filter (which the record must
2167 match) for the particular category. Filters are additive (meaning all
2168 rules must match to post that CDR).
2169 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
2170 module. Specifically, you may add additional columns into the table and
2171 they will be set, if you set the corresponding CDR variable name. Also,
2172 if you omit columns in your database table, they will be silently skipped
2173 (but a record will still be inserted, based on what columns remain). Note
2174 that the other two features from cdr_adaptive_odbc (alias and filter) are
2175 not currently supported.
2176 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
2177 has been disabled using the NoCDR application.
2179 Miscellaneous New Modules
2180 -------------------------
2181 * Added a new CDR module, cdr_sqlite3_custom.
2182 * Added a new realtime configuration module, res_config_sqlite
2183 * Added a new codec translation module, codec_resample, which re-samples
2184 signed linear audio between 8 kHz and 16 kHz to help support wideband
2186 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
2187 based on configuration templates that use Asterisk dialplan function and
2188 variable substitution. It should be possible to create phone profiles and
2189 templates that work for the majority of phones provisioned over http. It
2190 is currently only intended to provision a single user account per phone.
2191 An example profile and set of templates for Polycom phones is provided.
2192 NOTE: Polycom firmware is not included, but should be placed in
2193 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
2194 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
2195 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
2196 provided; there is a JACK() application, and a JACK_HOOK() function. Both
2197 interfaces create an input and output JACK port. The application makes
2198 these ports the endpoint of the call. The audio coming from the channel
2199 goes out the output port and whatever comes back in on the input port is
2200 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
2201 audiohook on the channel. This lets you run the audio coming from a
2202 channel through JACK, and whatever comes back in is what gets forwarded
2203 on as the channel's audio. This is very useful for building custom
2204 vocoders or doing recording or analysis of the channel's audio in another
2206 * Added a new module, res_config_curl, which permits using a HTTP POST url
2207 to retrieve, create, update, and delete realtime information from a remote
2208 web server. Note that this module requires func_curl.so to be loaded for
2209 backend functionality.
2210 * Added a new module, res_config_ldap, which permits the use of an LDAP
2211 server for realtime data access.
2212 * Added support for writing and running your dialplan in lua using the pbx_lua
2213 module. See configs/extensions.lua.sample for examples of how to do this.
2217 * Ability to use libcap to set high ToS bits when non-root
2218 on Linux. If configure is unable to find libcap then you
2219 can use --with-cap to specify the path.
2220 * Added maxfiles option to options section of asterisk.conf which allows you to specify
2221 what Asterisk should set as the maximum number of open files when it loads.
2222 * Added the jittertargetextra configuration option.
2223 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
2224 configuration files for the IP channel drivers. The new option is "cos".
2225 This information is also documented on the Asterisk wiki at
2226 https://wiki.asterisk.org/wiki/x/EYBG
2227 * When originating a call using AMI or pbx_spool that fails the reason for failure
2228 will now be available in the failed extension using the REASON dialplan variable.
2229 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
2230 It allows you to configure a prefix for auto-monitor recordings.
2231 * A new extension pattern matching algorithm, based on a trie, is introduced
2232 here, that could noticeably speed up mid-sized to large dialplans.
2233 It is NOT used by default, as duplicating the behaviour of the old pattern
2234 matcher is still under development. A config file option, in extensions.conf,
2235 in the [general] section, called "extenpatternmatchingnew", is by default
2236 set to false; setting that to true will force the use of the new algorithm.
2237 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
2238 be used to switch the algorithms at run time.
2239 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
2240 specifying which socket to use to connect to the running Asterisk daemon
2242 * Performance enhancements to the sched facility, which is used in
2243 the channel drivers, etc. Added hashtabs and doubly-linked lists
2244 to speed up deletion; start at the beginning or end of list to
2246 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
2247 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
2248 Added regression tests to the tests/ dir, also.
2249 * Added a refcount trace feature to astobj2 for those trying to balance
2250 object creation, deletion; work, play; space and time. See the
2251 notes in astobj2.h. Also, see utils/refcounter as well, as a
2252 quick way to find unbalanced refcounts in what could be a sea
2253 of objects that were balanced.
2254 * Added logging to 'make update' command. See update.log
2255 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
2256 do not come from the remote party.
2257 * Added the 'n' option to the SpeechBackground application to tell it to not
2258 answer the channel if it has not already been answered.
2259 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
2260 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
2262 * iLBC source code no longer included (see UPGRADE.txt for details)
2263 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
2264 deadlock is detected, a backtrace of the stack which led to the lock calls
2265 will be output to the CLI.
2266 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
2267 the "core show locks" CLI command will give lock information output as well
2268 as a backtrace of the stack which led to the lock calls.
2269 * users.conf now sports an optional alternateexts property, which permits
2270 allocation of additional extensions which will reach the specified user.
2271 * A new option for the configure script, --enable-internal-poll, has been added
2272 for use with systems which may have a buggy implementation of the poll system
2273 call. If you notice odd behavior such as the CLI being unresponsive on remote
2274 consoles, you may want to try using this option. This option is enabled by default
2275 on Darwin systems since it is known that the Darwin poll() implementation has
2279 --------------------
2280 * In addition to timing from DAHDI, there is a new timing module called
2281 res_timing_timerfd. In order to use this, you must be running Linux with
2282 a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
2283 script will be able to tell if you have the requirements. From menuselect, select
2284 res_timing_timerfd from the Resource Modules menu.
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