1 ==============================================================================
3 === This file documents the new and/or enhanced functionality added in
4 === the Asterisk versions listed below. This file does NOT include
5 === changes in behavior that would not be backwards compatible with
6 === previous versions; for that information see the UPGRADE.txt file
7 === and the other UPGRADE files for older releases.
9 ==============================================================================
11 ------------------------------------------------------------------------------
12 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
13 ------------------------------------------------------------------------------
17 * The expression parser now recognizes the ABS() absolute value function,
18 which will convert negative floating point values to positive values.
19 * The Asterisk build system will now build and install a shared library
20 (libasteriskssl.so) used to wrap various initialization and shutdown functions
21 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
22 that Asterisk can ensure that these functions do *not* get called by any
23 modules that are loaded into Asterisk, since they should only be called once
24 in any single process. If desired, this feature can be disabled by supplying
25 the "--disable-asteriskssl" option to the configure script.
26 * Threads belonging to a particular call are now linked with callids which get
27 added to any log messages produced by those threads. Log messages can now be
28 easily identified as involved with a certain call by looking at their call id.
29 This feature can be disabled in logger.conf with the display_callids option.
30 * The minimum DTMF duration can now be configured in asterisk.conf
31 as "mindtmfduration". The default value is (as before) set to 80 ms.
32 (previously it was only available in source code)
36 * mixmonitor list <channel> command will now show MixMonitor ID, and the filenames
37 of all running mixmonitors on a channel.
38 * The debuglevel of "pri set debug" is now a bitmask ranging from 0 to 15 if
39 numeric instead of 0, 1, or 2.
43 * Added menu action admin_toggle_mute_participants. This will mute / unmute
44 all non-admin participants on a conference. The confbridge configuration file
45 also allows for the default sounds played to all conference users when this
46 occurs to be overriden using sound_participants_unmuted and sound_participants_muted.
47 * Added menu action participant_count. This will playback the number of current
48 participants in a conference.
49 * Added announcement configuration option to user profile. If set the sound file will
50 be played to the user, and only the user, upon joining the conference bridge.
54 * Addition of the VM_INFO function - see Dialplan function changes
55 * The imapserver, imapport, and imapflags configuration options can now be
56 overriden on a user by user basis.
60 * Asterisk will no longer substitute CID number for CID name into display
61 name field if CID number exists without a CID name. This change improves
62 compatibility with certain device features such as Avaya IP500's directory
64 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
65 created using that setting to not be removed during SIP reload.
66 * Add support to realtime for the 'callbackextension' option
67 * When multiple peers exist with the same address, but differing
68 callbackextension options, incoming requests that are matched by address
69 will be matched to the peer with the matching callbackextension if it is
71 * NAT settings are now a combinable list of options. The equivalent of the
72 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
73 * Two new NAT options, auto_force_rport and auto_comedia, have been added
74 which set the force_rport and comedia options automatically if Asterisk
75 detects that an incoming SIP request crossed a NAT after being sent by
77 * Adds an option send_diversion which can be disabled to prevent
78 diversion headers from automatically being added to invites.
79 * Add support for lightweight NAT keepalive. If enabled a blank packet will
80 be sent to the remote host at a given interval to keep the NAT mapping open.
81 This can be enabled using the keepalive configuration option.
85 * Added a manager event "LocalBridge" for local channel call bridges between
86 the two pseudo-channels created.
90 * Added dialtone_detect option for analog ports to disconnect incoming
91 calls when dialtone is detected.
95 * Added ability to use multiple lines on phone, so for one device in
96 configuration multiple lines can be defined, it allows to have multiple calls
97 on one phone, callwaiting and switching between calls.
98 * Added option 'sharpdial' allowing end dialing by pressing # key
99 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
100 * Added global 'debug' option, that enables debug in channel driver
101 * Added ability for translation on-screen menu to multiple languages. Tested on
102 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
103 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
105 * Reworked dialing number input: added dialing by timeout, immediate dial on
106 on dialplan compare, phone number length now not limited by screen size
107 * Added ability for pickup a call using fetures.conf defined value and
112 * Codec lists may now be modified by the '!' character, to allow succinct
113 specification of a list of codecs allowed and disallowed, without the
114 requirement to use two different keywords. For example, to specify all
115 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
117 Music On Hold Changes
118 ---------------------
119 * Added 'announcement' option which will play at the start of MOH and between
120 songs in modes of MOH that can detect transitions between songs (eg.
125 * Added queue options autopausebusy and autopauseunavail for automatically
126 pausing a queue member when their device reports busy or congestion.
130 * When voicemail plays a message's envelope with saycid set to yes, when reaching
131 the caller id field it will play a recording of a file with the same base name
132 as the sender's callerid if there is a similarly named file in
133 <astspooldir>/recordings/callerids/
137 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
138 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
139 changed arguments to SayUnixTime so that every option is truly optional even
140 when using multiple options (so that j option could be used without having to
141 manually specify timezone and format) There are other beneftis eg. format can
142 now be used without specifying time zone as well.
143 * Added 'F()' option to Queue and Bridge. Similar to the dial option, these can
144 be supplied with arguments indicating where the callee should go after the caller
145 is hung up, or without options specified, the priority after the Queue/Bridge
147 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
148 channels respectively before the callee channels are called.
152 * New per parking lot options: comebackcontext and comebackdialtime. See
153 configs/features.conf.sample for more details.
155 * Channel variable PARKER is now set when comebacktoorigin is disabled in
158 * MixMonitor hooks now have IDs associated with them which can be used to assign
159 a target to StopMixMonitor. Use of MixMonitor's i(variable) option will allow
160 storage of the MixMontior ID in a channel variable. StopMixmonitor now accepts
161 that ID as an argument.
163 CDR postgresql driver changes
164 -----------------------------
165 * Added command "cdr show pgsql status" to check connection status
167 AMI (Asterisk Manager Interface) changes
168 ----------------------------------------
169 * Originate now generates an error response if the extension given
170 is not found in the dialplan
172 * MixMonitor will now show IDs associated with the mixmonitor upon creating them
173 if the i(variable) option is used. StopMixMonitor will accept MixMonitorID as
174 on option to close specific MixMonitors.
176 * The SIPshowpeer manager action response field "SIP-Forcerport" has been updated
177 to include information about peers configured with nat=auto_force_rport by
178 returning "A" if auto_force_rport is set and nat is detected, and "a" if it is
179 set and nat is not detected. "Y" and "N" are still returned if auto_force_rport
182 * Hangup now can take a regular expression as the Channel option. If you want
183 to hangup multiple channels, use /regex/ as the Channel option. Existing
184 behavior to hanging up a single channel is unchanged, but if you pass a regex,
185 the manager will send you a list of channels back that were hung up.
189 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
190 control of faxdetect.
194 * Allow the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to be
195 used within the dynamic weight attribute when specifying a mapping.
199 * Each logging destination and console now have an independent notion of the
200 current verbosity level. Logger.conf now allows an optional argument to
201 the 'verbose' specifier, indicating the level of verbosity sent to that
202 particular logging destination. Additionally, remote consoles now each
203 have their own verbosity level. The command 'core set verbose' will now set
204 a separate level for each remote console without affecting any other
209 * Addition of the VM_INFO function that can be used to retrieve voicemail
210 user information, such as the email address and full name.
211 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
213 * The REDIRECTING function now supports the redirecting original party id
215 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
216 lets you set some of the configuration options from the [general] section
217 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
218 the key sequence used to activate built-in features, such as blindxfer,
219 and automon. See the built-in documentation for details.
223 * A new option, 'I' has been added to app_followme.
224 By setting this option, Asterisk will not update the caller with
225 connected line changes when they occur. This is similar to app_dial
227 * The 'N' option is now ignored if the call is already answered.
231 * A new option, 'probation' has been added to rtp.conf
232 RTP in strictrtp mode can now require more than 1 packet to exit learning
233 mode with a new source (and by default requires 4). The probation option
234 allows the user to change the required number of packets in sequence to any
235 desired value. Use a value of 1 to essentially restore the old behavior.
236 Also, with strictrtp on, Asterisk will now drop all packets until learning
237 mode has successfully exited. These changes are based on how pjmedia handles
238 media sources and source changes.
242 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
243 instead of simply the uri. This is the format that MessageSend() can use
244 in the from parameter for outgoing SIP messages.
248 * A new module, res_corosync, has been introduced. This module uses the
249 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
250 of Asterisk servers to both Message Waiting Indication (MWI) and/or
251 Device State (presence) information. This module is very similar to, and
252 is a replacement for the res_ais module that was in previous releases of
257 * A new channel variable, AGIEXITONHANGUP, has been added which allows
258 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
259 AGI application would exit immediately after a channel hangup is detected.
260 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
261 are resolved and each address is attempted in turn until one succeeds or
264 ------------------------------------------------------------------------------
265 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
266 ------------------------------------------------------------------------------
270 * Asterisk now has protocol independent support for processing text messages
271 outside of a call. Messages are routed through the Asterisk dialplan.
272 SIP MESSAGE and XMPP are currently supported. There are options in
273 jabber.conf and sip.conf to allow enabling these features.
274 -> jabber.conf: see the "sendtodialplan" and "context" options.
275 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
276 and "outofcall_message_context" options.
277 The MESSAGE() dialplan function and MessageSend() application have been
278 added to go along with this functionality. More detailed usage information
279 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
280 * If real-time text support (T.140) is negotiated, it will be preferred for
281 sending text via the SendText application. For example, via SIP, messages
282 that were once sent via the SIP MESSAGE request would be sent via RTP if
283 T.140 text is negotiated for a call.
287 * parkedmusicclass can now be set for non-default parking lots.
289 Asterisk Manager Interface
290 --------------------------
291 * PeerStatus now includes Address and Port.
292 * Added Hold events for when the remote party puts the call on and off hold
293 for chan_dahdi ISDN channels.
294 * Added new action MeetmeListRooms to list active conferences (shows same
295 data as "meetme list" at the CLI).
296 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
297 Description field that is set by 'description' in the channel configuration
299 * Added Uniqueid header to UserEvent.
300 * Added new action FilterAdd to control event filters for the current session.
301 This requires the system permission and uses the same filter syntax as
302 filters that can be defined in manager.conf
303 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
304 versions had some instances of the event converted, but others were left
305 as-is. All Unlink events should now be converted to Bridge events. The AMI
306 protocol version number was incremented to 1.2 as a result of this change.
309 --------------------------
310 * The HTTP Server can bind to IPv6 addresses.
313 --------------------------
314 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
315 with busydetect. usage example: busypattern=200,200,200,600
318 --------------------------
319 * New 'gtalk show settings' command showing the current settings loaded from
321 * The 'logger reload' command now supports an optional argument, specifying an
322 alternate configuration file to use.
323 * 'dialplan add extension' command will now automatically create a context if
324 the specified context does not exist with a message indicated it did so.
325 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
326 Description field which can be populated with 'description' in the channel
327 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
330 --------------------------
331 * The filter option in cdr_adaptive_odbc now supports negating the argument,
332 thus allowing records which do NOT match the specified filter.
333 * Added ability to log CONGESTION calls to CDR
336 --------------------------
337 * Ability to define custom SILK formats in codecs.conf.
338 * Addition of speex32 audio format with translation.
339 * CELT codec pass-through support and ability to define
340 custom CELT formats in codecs.conf.
341 * Ability to read raw signed linear files with sample rates
342 ranging from 8khz - 192khz. The new file extensions introduced
343 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
344 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
345 Skinny, H.323, etc) can still only support the following codecs:
346 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
347 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
348 Video: h261, h263, h263p, h264, mpeg4
353 --------------------------
354 * New highly optimized and customizable ConfBridge application capable of
355 mixing audio at sample rates ranging from 8khz-96khz.
356 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
357 and bridge profiles on a channel.
358 * CONFBRIDGE_INFO dialplan function capable of retrieving information
359 about a conference such as locked status and number of parties, admins,
361 * Addition of video_mode option in confbridge.conf for adding video support
362 into a bridge profile.
363 * Addition of the follow_talker video_mode in confbridge.conf. This video
364 mode dynamically switches the video feed to always display the loudest talker
365 supplying video in the conference.
369 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
370 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
371 variables from asterisk.conf.
375 * Addition of the JITTERBUFFER dialplan function. This function allows
376 for jitterbuffering to occur on the read side of a channel. By using
377 this function conference applications such as ConfBridge and MeetMe can
378 have the rx streams jitterbuffered before conference mixing occurs.
379 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
381 * Added STRREPLACE function. This function let's the user search a variable
382 for a given string to replace with another string as many times as the
383 user specifies or just throughout the whole string.
384 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
385 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
386 * Added extensions to chan_ooh323 in function CHANNEL()
388 libpri channel driver (chan_dahdi) DAHDI changes
389 --------------------------
390 * Added moh_signaling option to specify what to do when the channel's bridged
391 peer puts the ISDN channel on hold.
392 * Added display_send and display_receive options to control how the display ie
393 is handled. To send display text from the dialplan use the SendText()
394 application when the option is enabled.
395 * Added mcid_send option to allow sending a MCID request on a span.
398 --------------------------
399 * Added setvar option to calendar.conf to allow setting channel variables on
400 notification channels.
401 * Added "calendar show types" CLI command to list registered calendar
405 --------------------------
406 * Added two new options, r and t with file name arguments to record
407 single direction (unmixed) audio recording separate from the bidirectional
408 (mixed) recording. The mixed file name argument is optional now as long
409 as at least one recording option is used.
412 --------------------------
413 * Added a new option, l, which will disable local call optimization for
414 channels involved with the FollowMe thread. Use this option to improve
415 compatability for a FollowMe call with certain dialplan apps, options, and
419 --------------------------
420 * Added option "k" that will automatically close the conference when there's
421 only one person left when a user exits the conference.
424 --------------------------
425 * cel_pgsql now supports the 'extra' column for data added using the
426 CELGenUserEvent() application.
429 --------------------------
430 * Support for defining hints has been added to pbx_lua. See the 'hints' table
431 in the sample extensions.lua file for syntax details.
432 * Applications that perform jumps in the dialplan such as Goto will now
433 execute properly. When pbx_lua detects that the context, extension, or
434 priority we are executing on has changed it will immediately return control
435 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
436 the priority after the currently executing priority.
437 * An autoservice is now started by default for pbx_lua channels. It can be
438 stopped and restarted using the autoservice_stop() and autoservice_start()
442 --------------------------
443 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
444 into a FAXStatus event with an 'Operation' header that will be either
445 'send', 'receive', and 'gateway'.
446 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
447 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
448 feature will handle converting a fax call between an audio T.30 fax terminal
449 and an IFP T.38 fax terminal.
453 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
454 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
455 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
459 * Added general option negative_penalty_invalid default off. when set
460 members are seen as invalid/logged out when there penalty is negative.
461 for realtime members when set remove from queue will set penalty to -1.
462 * Added queue option autopausedelay when autopause is enabled it will be
463 delayed for this number of seconds since last successful call if there
464 was no prior call the agent will be autopaused immediately.
465 * Added member option ignorebusy this when set and ringinuse is not
466 will allow per member control of multiple calls as ringinuse does for
468 * Added global option check_state_unknown to enforce checking of device state
469 when the device state is unknown app_queue will see unknown as available.
473 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
475 * Added 'k' option to MeetMe to automatically kill the conference when there's only
476 one participant left (much like a normal call bridge)
477 * Added extra argument to Originate to set timeout.
481 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
482 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
483 utility in the UTILS section of menuselect. If an existing astdb is found and no
484 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
485 convert an existing astdb to the SQLite3 version automatically at runtime.
489 * Modules marked as deprecated are no longer marked as building by default. Enabling
490 these modules is still available via menuselect.
494 * authdebug is now disabled by default. To enable this functionaility again
495 set authdebug = yes in iax.conf.
499 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
500 releases it was disabled.
504 * The PBX core previously made a call with a non-existing extension test for
505 extension s@default and jump there if the extension existed.
506 This was a bad default behaviour and violated the principle of least surprise.
507 It has therefore been changed in this release. It may affect some
508 applications and configurations that rely on this behaviour. Most channel
509 drivers have avoided this for many releases by testing whether the extension
510 called exists before starting the PBX and generating a local error.
511 This behaviour still exists and works as before.
513 Extension "s" is used when no extension is given in a channel driver,
514 like immediate answer in DAHDI or calling to a domain with no user part
517 ------------------------------------------------------------------------------
518 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
519 ------------------------------------------------------------------------------
523 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
524 now defaults to force_rport. It is very important that phones requiring nat=no be
525 specifically set as such instead of relying on the default setting. If at all
526 possible, all devices should have nat settings configured in the general section as
527 opposed to configuring nat per-device.
528 * Added preferred_codec_only option in sip.conf. This feature limits the joint
529 codecs sent in response to an INVITE to the single most preferred codec.
530 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
531 to be used for the outgoing call. It must be one of the codecs configured
533 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
534 to be used for holding a private key. If tlsprivatekey is not specified,
535 tlscertfile is searched for both public and private key.
536 * Added tlsclientmethod option to sip.conf. This allows the protocol for
537 outbound client connections to be specified.
538 * The sendrpid parameter has been expanded to include the options
539 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
540 header to be sent (equivalent to setting sendrpid=yes) and setting
541 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
542 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
543 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
544 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
545 will accept the SDP even if the SDP version number is not properly incremented,
546 but will generate a warning in the log indicating that the SIP peer that sent
547 the SDP should have the 'ignoresdpversion' option set.
548 * The 'nat' option has now been been changed to have yes, no, force_rport, and
549 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
550 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
551 remote side requests it and disables symmetric RTP support. Setting it to
552 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
553 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
554 and enables symmetric RTP support.
555 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
556 response. This permits the master channel to know how each channel dialled
557 in a multi-channel setup resolved in an individual way. This carries a
558 performance penalty and can be disabled in sip.conf using the
559 'storesipcause' option.
560 * Added 'externtcpport' and 'externtlsport' options to allow custom port
561 configuration for the externip and externhost options when tcp or tls is used.
562 * Added support for message body (stored in content variable) to SIP NOTIFY message
563 accessible via AMI and CLI.
564 * Added 'media_address' configuration option which can be used to explicitly specify
565 the IP address to use in the SDP for media (audio, video, and text) streams.
566 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
567 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
569 * Added 'use_q850_reason' configuration option for generating and parsing
570 if available Reason: Q.850;cause=<cause code> header. It is implemented
571 in some gateways for better passing PRI/SS7 cause codes via SIP.
572 * When dialing SIP peers, a new component may be added to the end of the dialstring
573 to indicate that a specific remote IP address or host should be used when dialing
574 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
575 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
576 ability to selectively force bridged channels to also be encrypted is also
577 implemented. Branching in the dialplan can be done based on whether or not
578 a channel has secure media and/or signaling.
579 * Added directmediapermit/directmediadeny to limit which peers can send direct media
581 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
582 Charge messages to snom phones.
583 * Added support for G.719 media streams.
584 * Added support for 16khz signed linear media streams.
585 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
586 RTP has been outfitted with the same abilities.
587 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
588 available in device configurations as well as in the dial plan.
589 * Addition of the 'subscribe_network_change' option for turning on and off
590 res_stun_monitor module support in chan_sip.
591 * Addition of the 'auth_options_requests' option for turning on and off
592 authentication for OPTIONS requests in chan_sip.
596 * Add #tryinclude statement for config files. This provides the same
597 functionality as the #include statement however an asterisk module will
598 still load if the filename does not exist. Using the #include statement
599 Asterisk will not allow the module to load.
603 * Added rtsavesysname option into iax.conf to allow the systname to be saved
605 * Added the ability for chan_iax2 to inform the dialplan whether or not
606 encryption is being used. This interoperates with the SIP SRTP implementation
607 so that a secure SIP call can be bridged to a secure IAX call when the
608 dialplan requires bridged channels to be "secure".
609 * Addition of the 'subscribe_network_change' option for turning on and off
610 res_stun_monitor module support in chan_iax.
615 * Added ability to preset channel variables on indicated lines with the setvar
616 configuration option. Also, clearvars=all resets the list of variables back
618 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
619 See configs/res_pktccops.conf for more information.
621 XMPP Google Talk/Jingle changes
622 -------------------------------
623 * Added the externip option to gtalk.conf.
624 * Added the stunaddr option to gtalk.conf which allows for the automatic
625 retrieval of the external ip from a stun server.
629 * Added 'p' option to PickupChan() to allow for picking up channel by the first
630 match to a partial channel name.
631 * Added .m3u support for Mp3Player application.
632 * Added progress option to the app_dial D() option. When progress DTMF is
633 present, those values are sent immediately upon receiving a PROGRESS message
634 regardless if the call has been answered or not.
635 * Added functionality to the app_dial F() option to continue with execution
636 at the current location when no parameters are provided.
637 * Added the 'a' option to app_dial to answer the calling channel before any
638 announcements or macros are executed.
639 * Modified app_dial to set answertime when the called channel answers even if
640 the called channel hangs up during playback of an announcement.
641 * Modified app_dial 'r' option to support an additional parameter to play an
642 indication tone from indications.conf
643 * Added c() option to app_chanspy. This option allows custom DTMF to be set
644 to cycle through the next available channel. By default this is still '*'.
645 * Added x() option to app_chanspy. This option allows DTMF to be set to
646 exit the application.
647 * The Voicemail application has been improved to automatically ignore messages
648 that only contain silence.
649 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
650 associated mailbox(es) to be greetings-only.
651 * The ChanSpy application now has the 'S' option, which makes the application
652 automatically exit once it hits a point where no more channels are available
654 * The ChanSpy application also now has the 'E' option, which spies on a single
655 channel and exits when that channel hangs up.
656 * The MeetMe application now turns on the DENOISE() function by default, for
657 each participant. In our tests, this has significantly decreased background
658 noise (especially noisy data centers).
659 * Voicemail now permits storage of secrets in a separate file, located in the
660 spool directory of each individual user. The control for this is located in
661 the "passwordlocation" option in voicemail.conf. Please see the sample
662 configuration for more information.
663 * The ChanIsAvail application now exposes the returned cause code using a separate
664 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
665 * Added 'd' option to app_followme. This option disables the "Please hold"
667 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
668 received will terminate recording.
669 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
670 Previously the folder could only be set per context, but has now been extended
671 using the imapfolder option.
672 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
673 * Voicemail now allows the pager date format to be specified separately from the
675 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
676 to allow joining, leaving, and sending text to group chats.
677 * MeetMe has a new option 'G' to play an announcement before joining a conference.
678 * Page has a new option 'A(x)' which will playback an announcement simultaneously
679 to all paged phones (and optionally excluding the caller's one using the new
680 option 'n') before the call is bridged.
681 * The 'f' option to Dial has been augmented to take an optional argument. If no
682 argument is provided, the 'f' option works as it always has. If an argument is
683 provided, then the connected party information of all outgoing channels created
684 during the Dial will be set to the argument passed to the 'f' option.
685 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
687 * The OSP lookup application adds in/outbound network ID, optional security,
688 number portability, QoS reporting, destination IP port, custom info and service
690 * Added new application VMSayName that will play the recorded name of the voicemail
691 user if it exists, otherwise will play the mailbox number.
692 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
693 retrieve state for a particular bridge, where <name> is the conference name
694 * app_directory now allows exiting at any time using the operator or pound key.
695 * Voicemail now supports setting a locale per-mailbox.
696 * Two new applications are provided for declining counting phrases in multiple
697 languages. See the application notes for SayCountedNoun and SayCountedAdj for
699 * Voicemail now runs the externnotify script when pollmailboxes is activated and
701 * Voicemail now includes rdnis within msgXXXX.txt file.
702 * ExternalIVR now supports IPv6 addresses.
703 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
704 at https://wiki.asterisk.org/wiki/x/oQBB
705 * ParkedCall and Park can now specify the parking lot to use.
709 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
710 over SRV records associated with a specific service. From the CLI, type
711 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
712 details on how these may be used.
713 * PITCH_SHIFT dialplan function added. This function can be used to modify the
714 pitch of a channel's tx and rx audio streams.
715 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
716 setting various connected line and redirecting party information.
717 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
718 support ISDN subaddressing.
719 * The CHANNEL() function now supports the "name" and "checkhangup" options.
720 * For DAHDI channels, the CHANNEL() dialplan function now allows
721 the dialplan to request changes in the configuration of the active
722 echo canceller on the channel (if any), for the current call only.
725 exten => s,n,Set(CHANNEL(echocan_mode)=off)
727 The possible values are:
729 on - normal mode (the echo canceller is actually reinitialized)
731 fax - FAX/data mode (NLP disabled if possible, otherwise completely
733 voice - voice mode (returns from FAX mode, reverting the changes that
734 were made when FAX mode was requested)
735 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
736 and setting variables on the channel which created the current channel.
737 Administrators should take care to avoid naming conflicts, when multiple
738 channels are dialled at once, especially when used with the Local channel
739 construct (which all could set variables on the master channel). Usage
740 of the HASH() dialplan function, with the key set to the name of the slave
741 channel, is one approach that will avoid conflicts.
742 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
744 * func_odbc now allows multiple row results to be retrieved without using
745 mode=multirow. If rowlimit is set, then additional rows may be retrieved
746 from the same query by using the name of the function which retrieved the
747 first row as an argument to ODBC_FETCH().
748 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
749 dialplan. This function returns the content of the received message.
750 * Added REPLACE, which searches a given variable name for a set of characters,
751 then either replaces them with a single character or deletes them.
752 * Added PASSTHRU, which literally passes the same argument back as its return
753 value. The intent is to be able to use a literal string argument to
754 functions that currently require a variable name as an argument.
755 * HASH-associated variables now can be inherited across channel creation, by
756 prefixing the name of the hash at assignment with the appropriate number of
757 underscores, just like variables.
758 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
759 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
760 whether or not channels that are bridged to the current channel will be
761 required to have secure signaling and/or media.
762 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
763 the current channel has secure signaling and/or media.
764 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
765 "no_media_path" option.
766 Returns "0" if there is a B channel associated with the call.
767 Returns "1" if no B channel is associated with the call. The call is either
768 on hold or is a call waiting call.
769 * Added option to dialplan function CDR(), the 'f' option
770 allows for high resolution times for billsec and duration fields.
771 * FILE() now supports line-mode and writing.
772 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
773 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
777 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
778 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
779 and is set when a dynamic feature is triggered.
780 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
781 to dynamically create a new parking lot matching the value this varible is
783 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
784 features.conf that should be the base for dynamic parkinglots.
785 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
786 parkinglot should have.
787 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
788 parkinglot should have.
789 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
794 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
796 * Added 'R' option to app_queue. This option stops moh and indicates ringing
797 to the caller when an Agent's phone is ringing. This can be used to indicate
798 to the caller that their call is about to be picked up, which is nice when
799 one has been on hold for an extened period of time.
800 * A new config option, penaltymemberslimit, has been added to queues.conf.
801 When set this option will disregard penalty settings when a queue has too
803 * A new option, 'I' has been added to both app_queue and app_dial.
804 By setting this option, Asterisk will not update the caller with
805 connected line changes or redirecting party changes when they occur.
806 * A 'relative-periodic-announce' option has been added to queues.conf. When
807 enabled, this option will cause periodic announce times to be calculated
808 from the end of announcements rather than from the beginning.
809 * The autopause option in queues.conf can be passed a new value, "all." The
810 result is that if a member becomes auto-paused, he will be paused in all
811 queues for which he is a member, not just the queue that failed to reach
813 * Added dialplan function QUEUE_EXISTS to check if a queue exists
814 * The queue logger now allows events to optionally propagate to a file,
815 even when realtime logging is turned on. Additionally, realtime logging
816 supports sending the event arguments to 5 individual fields, although it
817 will fallback to the previous data definition, if the new table layout is
820 mISDN channel driver (chan_misdn) changes
821 ----------------------------------------
822 * Added display_connected parameter to misdn.conf to put a display string
823 in the CONNECT message containing the connected name and/or number if
824 the presentation setting permits it.
825 * Added display_setup parameter to misdn.conf to put a display string
826 in the SETUP message containing the caller name and/or number if the
827 presentation setting permits it.
828 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
829 indicate the dialplan settings are to be obtained from the asterisk
831 * Made misdn.conf parameter callerid accept the "name" <number> format
832 used by the rest of the system.
833 * Made use the nationalprefix and internationalprefix misdn.conf
834 parameters to prefix any received number from the ISDN link if that
835 number has the corresponding Type-Of-Number. NOTE: This includes
836 comparing the incoming call's dialed number against the MSN list.
837 * Added the following new parameters: unknownprefix, netspecificprefix,
838 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
839 received number from the ISDN link if that number has the corresponding
841 * Added new dialplan application misdn_command which permits controlling
842 the CCBS/CCNR functionality.
843 * Added new dialplan function mISDN_CC which permits retrieval of various
844 values from an active call completion record.
845 * For PTP, you should manually send the COLR of the redirected-to party
846 for an incomming redirected call if the incoming call could experience
847 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
848 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
849 if the REDIRECTING(from-num) is not empty.
850 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
851 option on all of the REDIRECTING statements before dialing the
852 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
853 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
854 redirecting-to presentation (COLR) when it becomes available.
855 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
858 thirdparty mISDN enhancements
859 -----------------------------
860 mISDN has been modified by Digium, Inc. to greatly expand facility message
862 * Enhanced COLP support for call diversion and transfer.
865 The latest modified mISDN v1.1.x based version is available at:
866 http://svn.digium.com/svn/thirdparty/mISDN/trunk
867 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
869 Tagged versions of the modified mISDN code are available under:
870 http://svn.digium.com/svn/thirdparty/mISDN/tags
871 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
873 libpri channel driver (chan_dahdi) DAHDI changes
874 -------------------------------------------
875 * The channel variable PRIREDIRECTREASON is now just a status variable
876 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
877 to read and alter the reason.
878 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
879 redirected-to party for an incomming redirected call if the incoming call
880 could experience further redirects. Just set the
881 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
882 to the COLR. A call has been redirected if the REDIRECTING(count) is not
884 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
885 use the inhibit(i) option on all of the REDIRECTING statements before
886 dialing the redirected-to party. You still have to set the
887 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
888 will update the redirecting-to presentation (COLR) when it becomes available.
889 * Added the ability to ignore calls that are not in a Multiple Subscriber
890 Number (MSN) list for PTMP CPE interfaces.
891 * Added dynamic range compression support for dahdi channels. It is
892 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
893 * Added support for ISDN calling and called subaddress with partial support
894 for connected line subaddress.
895 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
896 * Added handling of received HOLD/RETRIEVE messages and the optional ability
897 to transfer a held call on disconnect similar to an analog phone.
898 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
899 Will reroute/deflect an outgoing call when receive the message.
900 Can use the DAHDISendCallreroutingFacility to send the message for the
902 * Added standard location to add options to chan_dahdi dialing:
903 Dial(DAHDI/g1[/extension[/options]])
906 R Reverse charging indication
907 * Added Reverse Charging Indication (Collect calls) send/receive option.
908 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
909 Dial(DAHDI/g1/extension/R)
910 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
911 (requires latest LibPRI)
912 * Added ability to send/receive keypad digits in the SETUP message.
913 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
914 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
915 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
916 (requires latest LibPRI)
917 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
918 to eliminate tromboned calls. A tromboned call goes out an interface and comes
919 back into the same interface. Tromboned calls happen because of call routing,
920 call deflection, call forwarding, and call transfer.
921 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
922 * Added the ability to support call waiting calls. (The SETUP has no B channel
924 * Added Malicious Call ID (MCID) event to the AMI call event class.
925 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
927 Asterisk Manager Interface
928 --------------------------
929 * The Hangup action now accepts a Cause header which may be used to
930 set the channel's hangup cause.
931 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
932 to specify a separate .pem file to hold a private key. By default sslcert
933 is used to hold both the public and private key.
934 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
935 for options containing the 'tls' prefix. For example, 'sslenable' is now
936 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
937 across all .conf files. All affected sample.conf files have been modified to
938 reflect this change. Previous options such as 'sslenable' still work,
939 but options with the 'tls' prefix are preferred.
940 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
941 in a channel. (res_mutestream.so)
942 * The configuration file manager.conf now supports a channelvars option, which
943 specifies a list of channel variables to include in each channel-oriented
945 * The redirect command now has new parameters ExtraContext, ExtraExtension,
946 and ExtraPriority to allow redirecting the second channel to a different
947 location than the first.
948 * Added new event "JabberStatus" in the Jabber module to monitor buddies
950 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
951 in a MixMonitor recording.
952 * The 'iax2 show peers' output is now similar to the expected output of
954 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
956 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
957 AOC-E messages on a channel.
958 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
959 conform more closely to similar events.
960 * Added a new eventfilter option per user to allow whitelisting and blacklisting
962 * Added optional parkinglot variable for park command.
963 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
964 if CallerIDNum and CallerIDName headers are also present.
966 Channel Event Logging
967 ---------------------
968 * A new interface, CEL, is introduced here. CEL logs single events, much like
969 the AMI, but it differs from the AMI in that it logs to db backends much
970 like CDR does; is based on the event subsystem introduced by Russell, and
971 can share in all its benefits; allows multiple backends to operate like CDR;
972 is specialized to event data that would be of concern to billing sytems,
973 like CDR. Backends for logging and accounting calls have been produced,
974 but a new CDR backend is still in development.
978 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
979 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
980 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
981 * Multiple files and formats can now be specified in cdr_custom.conf.
982 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
983 See configs/cdr_syslog.conf.sample for more information.
984 * A 'sequence' field has been added to CDRs which can be combined with
985 linkedid or uniqueid to uniquely identify a CDR.
986 * Handling of billsec and duration field has changed. If your table definition
987 specifies those fields as float,double or similar they will now be logged with
988 microsecond accuracy instead of a whole integer.
990 Calendaring for Asterisk
991 ------------------------
992 * A new set of modules were added supporing calendar integration with Asterisk.
993 Dialplan functions for reading from and writing to calendars are included,
994 as well as the ability to execute dialplan logic upon calendar event notifications.
995 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
996 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
997 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
998 2003 support does not support forms-based authentication).
1000 Call Completion Supplementary Services for Asterisk
1001 ---------------------------------------------------
1002 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
1003 DAHDI/ISDN supports call completion for the following switch types:
1004 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
1005 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
1007 Multicast RTP Support
1008 ---------------------
1009 * A new RTP engine and channel driver have been added which supports Multicast RTP.
1010 The channel driver can be used with the Page application to perform multicast RTP
1011 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
1012 Type can be either basic or linksys.
1013 Destination is the IP address and port for the RTP packets.
1014 Control address is specific to the linksys type and is used for sending the control
1015 packets unique to them.
1017 Security Events Framework
1018 -------------------------
1019 * Asterisk has a new C API for reporting security events. The module res_security_log
1020 sends these events to the "security" logger level. Currently, AMI is the only
1021 Asterisk component that reports security events. However, SIP support will be
1022 coming soon. For more information on the security events framework, see the
1023 "Asterisk Security Framework" section of the Asterisk wiki at
1024 https://wiki.asterisk.org/wiki/x/wgBQ
1025 * SIP support was added in Asterisk 10
1026 * This API now supports IPv6 addresses
1030 * A technology independent fax frontend (res_fax) has been added to Asterisk.
1031 * A spandsp based fax backend (res_fax_spandsp) has been added.
1032 * The app_fax module has been deprecated in favor of the res_fax module and
1033 the new res_fax_spandsp backend.
1034 * The SendFAX and ReceiveFAX applications now send their log messages to a
1035 'fax' logger level, instead of to the generic logger levels. To see these
1036 messages, the system's logger.conf file will need to direct the 'fax' logger
1037 level to one or more destinations; the logger.conf.sample file includes an
1038 example of how to do this. Note that if the 'fax' logger level is *not*
1039 directed to at least one destination, log messages generated by these
1040 applications will be lost, and that if the 'fax' logger level is directed to
1041 the console, the 'core set verbose' and 'core set debug' CLI commands will
1042 have no effect on whether the messages appear on the console or not.
1046 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
1047 Now, in order to enable transmitting silence during record the transmit_silence
1048 option should be used. transmit_silence_during_record remains a valid option, but
1049 defaults to the behavior of the transmit_silence option.
1050 * Addition of the Unit Test Framework API for managing registration and execution
1051 of unit tests with the purpose of verifying the operation of C functions.
1052 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
1053 XMPP text messages to the remote JID.
1054 * Modules.conf has a new option - "require" - that marks a module as critical for
1055 the execution of Asterisk.
1056 If one of the required modules fail to load, Asterisk will exit with a return
1058 * An 'X' option has been added to the asterisk application which enables #exec support.
1059 This allows #exec to be used in asterisk.conf.
1060 * jabber.conf supports a new option auth_policy that toggles auto user registration.
1061 * A new lockconfdir option has been added to asterisk.conf to protect the
1062 configuration directory (/etc/asterisk by default) during reloads.
1063 * The parkeddynamic option has been added to features.conf to enable the creation
1064 of dynamic parkinglots.
1065 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
1066 the reportalarms config option.
1067 * chan_dahdi supports dialing configuring and dialing by device file name.
1068 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
1069 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
1070 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
1071 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
1072 Handy for the above name-based syntax as it does not depend on
1073 initialization order.
1074 * The Realtime dialplan switch now caches entries for 1 second. This provides a
1075 significant increase in performance (about 3X) for installations using this switchtype.
1076 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
1077 AIS. For more information, please see the Distributed Device State section of the
1078 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1079 * The addition of G.719 pass-through support.
1080 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
1081 during device configuration.
1082 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
1083 have less than 3 lines on the LCD.
1084 * Realtime now supports database failover. See the sample extconfig.conf for details.
1085 * The addition of improved translation path building for wideband codecs. Sample
1086 rate changes during translation are now avoided unless absolutely necessary.
1087 * The addition of the res_stun_monitor module for monitoring and reacting to network
1088 changes while behind a NAT.
1092 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
1093 optionally accept a filename, to apply the setting only to the code generated from
1094 that source file when Asterisk was built. However, there are some modules in Asterisk
1095 that are composed of multiple source files, so this did not result in the behavior
1096 that users expected. In this version, 'core set debug' and 'core set verbose'
1097 can optionally accept *module* names instead (with or without the .so extension),
1098 which applies the setting to the entire module specified, regardless of which source
1099 files it was built from.
1100 * New 'manager show settings' command showing the current settings loaded from
1102 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
1103 the channel hangup request to all channels.
1104 * Added a "core reload" CLI command that executes a global reload of Asterisk.
1106 ------------------------------------------------------------------------------
1107 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
1108 ------------------------------------------------------------------------------
1112 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
1113 Snom phones use this for call pickup of extensions that the phone is
1115 * Added support for setting the domain in the URI for caller of an
1116 outbound call by using the SIPFROMDOMAIN channel variable.
1117 * Added a new configuration option "remotesecret" for authentication to
1118 remote services. For backwards compatibility, "secret" still has the
1119 same function as before, but now you can configure both a remote secret and a
1120 local secret for mutual authentication.
1121 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
1122 the sound will be played to the target of an attended transfer
1123 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
1124 finer control over how many peers Asterisk will qualify and the gap between them
1125 when all peers need to be qualified at the same time.
1126 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
1127 (either globally or for a specific peer), chan_sip will treat any SDP data
1128 it receives as new data and update the media stream accordingly. By
1129 default, Asterisk will only modify the media stream if the SDP session
1130 version received is different from the current SDP session version. This
1131 option is required to interoperate with devices that have non-standard SDP
1132 session version implementations (observed with Microsoft OCS). This option
1133 is disabled by default.
1134 * The parsing of register => lines in sip.conf has been modified to allow a port
1135 to be present in the "user" portion. Please see the sip.conf.sample file for more
1137 * Added support for subscribing to MWI on a remote server and making the status available
1138 as a mailbox. Please see the sip.conf.sample file for more information.
1139 * Added a function to remove SIP headers added in the dialplan before the
1140 first INVITE is generated - SIPRemoveHeader()
1141 * Channel variables set with setvar= in a device configuration is now
1142 set both for inbound and outbound calls.
1143 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
1147 * Added immediate option to iax.conf
1148 * Added forceencryption option to iax.conf
1149 * Added Encryption and Trunk status to manager command "iaxpeers"
1153 * The configuration file now holds separate sections for devices and lines.
1154 Please have a look at configs/skinny.conf.sample and change your skinny.conf
1159 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
1160 support for LibOpenR2. http://www.libopenr2.org/
1161 * The UK option waitfordialtone has been added for use with BT analog
1163 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
1164 is used in conjunction with the 'faxdetect' configuration option. When
1165 'faxbuffers' is used and fax tones are detected, the channel will dynamically
1166 switch to the configured faxbuffers policy. For example, to use 6 buffers
1167 and a 'full' buffer policy for a fax transmission, add:
1169 The faxbuffers configuration will be in affect until the call is torn down.
1170 * Added service message support for 4ESS/5ESS switches.
1174 * For DAHDI channels, the CHANNEL() dialplan function now
1175 supports changing the channel's buffer policy (for the current
1176 call only), using this syntax:
1178 exten => s,n,Set(CHANNEL(buffers)=6,full)
1180 This would change the channel to the 'full' buffer policy and
1181 6 (six) buffers. Possible options for this setting are the same
1182 as those in chan_dahdi.conf.
1183 * Added a new dialplan function, CURLOPT, which permits setting various
1184 options that may be useful with the CURL dialplan function, such as
1185 cookies, proxies, connection timeouts, passwords, etc.
1186 * Permit the syntax and synopsis fields of the corresponding dialplan
1187 functions to be individually set from func_odbc.conf.
1188 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
1189 * func_odbc now may specify an insert query to execute, when the write query
1190 affects 0 rows (usually indicating that no such row exists).
1191 * Added a new dialplan function, LISTFILTER, which permits removing elements
1192 from a set list, by name. Uses the same general syntax as the existing CUT
1193 and FIELDQTY dialplan functions, which also manage lists.
1194 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
1195 obtaining realtime data from the dialplan.
1196 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
1197 a subroutine when using the GoSub() and Return() applications.
1198 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
1199 of "core show function AUDIOHOOK_INHERIT" from the CLI
1200 * Added AES_ENCRYPT. For information on its use, please see the output
1201 of "core show function AES_ENCRYPT" from the CLI
1202 * Added AES_DECRYPT. For information on its use, please see the output
1203 of "core show function AES_DECRYPT" from the CLI
1204 * func_odbc now supports database transactions across multiple queries.
1208 * Scheduled meetme conferences may now have their end times extended by
1210 * app_authenticate now gives the ability to select a prompt other than
1212 * app_directory now pays attention to the searchcontexts setting in
1213 voicemail.conf and will look through all contexts, if no context is
1214 specified in the initial argument.
1215 * A new application, Originate, has been introduced, that allows asynchronous
1216 call origination from the dialplan.
1217 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
1218 in addition to the setting in the "general" context.
1219 * Added ConfBridge dialplan application which does conference bridges without
1220 DAHDI. For information on its use, please see the output of
1221 "core show application ConfBridge" from the CLI.
1225 * The Asterisk CLI has a new command, "channel redirect", which is similar in
1226 operation to the AMI Redirect action.
1227 * extensions.conf now allows you to use keyword "same" to define an extension
1228 without actually specifying an extension. It uses exactly the same pattern
1229 as previously used on the last "exten" line. For example:
1230 exten => 123,1,NoOp(something)
1231 same => n,SomethingElse()
1232 * musiconhold.conf classes of type 'files' can now use relative directory paths,
1233 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
1234 * All deprecated CLI commands are removed from the sourcecode. They are now handled
1235 by the new clialiases module. See cli_aliases.conf.sample file.
1236 * Times within timespecs are now accurate down to the minute. This is a change
1237 from historical Asterisk, which only provided timespecs rounded to the nearest
1238 even (read: evenly divisible by 2) minute mark.
1239 * The realtime switch now supports an option flag, 'p', which disables searches for
1241 * In addition to a time range and date range, timespecs now accept a 5th optional
1242 argument, timezone. This allows you to perform time checks on alternate
1243 timezones, especially if those daylight savings time ranges vary from your
1244 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
1246 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
1247 give you the correct output for an asterisk box behind nat. It will give you the
1248 externhost and localnet settings.
1249 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
1250 can connect calls in passthrough mode, as well as record and play back files.
1251 * Successful and unsuccessful call pickup can now be alerted through sounds, by
1252 using pickupsound and pickupfailsound in features.conf.
1253 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
1254 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
1255 instead of the /var/run/asterisk.pid where it used to be. This will make
1256 installs as non-root easier to manage.
1261 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
1262 be written; they will no longer be explicitly written.
1264 Asterisk Manager Interface
1265 --------------------------
1266 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
1267 a non-empty value) in your request. If you do this, any pending AMI events will
1268 *not* be included in the response to your request as they would normally, but
1269 will be left in the event queue for the next request you make to retrieve. For
1270 some applications, this will allow you to guarantee that you will only see
1271 events in responses to 'WaitEvent' actions, and can better know when to expect them.
1272 To know whether the Asterisk server supports this header or not, your client can
1273 inspect the first response back from the server to see if it includes this header:
1275 Pragma: SuppressEvents
1277 If this is included, the server supports event suppression.
1279 * Added 4 new Actions to list skinny device(s) and line(s)
1285 LDAP Schema File Additions
1286 --------------------------
1287 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
1288 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
1290 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
1291 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
1292 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
1293 * Removed redundant IPaddr (there's already IPAddress)
1294 - Gives more configuration Flags for SIP-Users available (tested)
1295 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
1296 without extensibleObject (which really should be the last resort); gives
1297 also additional possibilities for LDAP-filter
1299 ------------------------------------------------------------------------------
1300 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
1301 ------------------------------------------------------------------------------
1303 Device State Handling
1304 ---------------------
1305 * The event infrastructure in Asterisk got another big update to help support
1306 distributed events. It currently supports distributed device state and
1307 distributed Voicemail MWI (Message Waiting Indication). A new module has
1308 been merged, res_ais, which facilitates communicating events between servers.
1309 It uses the SAForum AIS (Service Availability Forum Application Interface
1310 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
1311 a cluster of Asterisk servers, and to share events between them. For more
1312 information on setting this up, refer to the Distributed Device State section
1313 of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1317 * Added a new dialplan function, AST_CONFIG(), which allows you to access
1318 variables from an Asterisk configuration file.
1319 * The JACK_HOOK function now has a c() option to supply a custom client name.
1320 * Added two new dialplan functions from libspeex for audio gain control and
1321 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
1322 rx directions of a channel from the dialplan.
1323 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
1324 based on other parameters. The default is still to search based on the
1325 forwarding station ID. However, there are new options that allow you to search
1326 based on the message desk terminal ID, or the message desk number.
1327 * TIMEOUT() has been modified to be accurate down to the millisecond.
1328 * ENUM*() functions now include the following new options:
1329 - 'u' returns the full URI and does not strip off the URI-scheme.
1330 - 's' triggers ISN specific rewriting
1331 - 'i' looks for branches into an Infrastructure ENUM tree
1332 - 'd' for a direct DNS lookup without any flipping of digits.
1333 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
1334 * CHANNEL() now has options for the maximum, minimum, and standard or normal
1335 deviation of jitter, rtt, and loss for a call using chan_sip.
1337 DAHDI channel driver (chan_dahdi) Changes
1338 ----------------------------------------
1339 * Channels can now be configured using named sections in chan_dahdi.conf, just
1340 like other channel drivers, including the use of templates.
1341 * The default for pridialplan has changed from 'national' to 'unknown'.
1345 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
1346 to something that matches the pattern a hint will be created using the contents
1347 and variables evaluated.
1348 * Dialplan matching has been extended to allow an extension to return to the
1349 PBX core to wait for more digits. This is done by using the new dialplan
1350 application called "Incomplete". This will permit a whole new level of
1351 extension control, by giving the administrator more control over early
1352 matches employing one of the short-circuit pattern match operators. Note
1353 that custom applications can trigger this same behavior by returning the
1354 special value AST_PBX_INCOMPLETE.
1358 * Directory now permits both first and last names to be matched at the same
1359 time. In addition, the number of digits to enter of the name can be set in
1360 the arguments to Directory; previously, you could enter only 3, regardless
1361 of how many names are in your company. For large companies, this should be
1363 * Voicemail now permits a mailbox setting to wrap around from first to last
1364 messages, if the "messagewrap" option is set to a true value.
1365 * Voicemail now permits an external script to be run, for password validation.
1366 The script should output "VALID" or "INVALID" on stdout, depending upon the
1367 wish to validate or invalidate the password given. Arguments are:
1368 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
1370 * Dial has a new option: F(context^extension^pri), which permits a callee to
1371 continue in the dialplan, at the specified label, if the caller hangs up.
1372 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
1373 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
1374 * The Jack application now has a c() option to supply a custom client name.
1375 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
1376 like the pre-existing whisper mode, except that the spy can also talk to the
1377 participant on the bridged channel as well.
1378 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
1379 to be spoken instead of the channel name or number. For more information on the
1380 use of this option, issue the command "core show application ChanSpy" from the
1382 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
1383 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
1384 words, if using the 'd' option, it is not possible to enter a number to append to
1385 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
1386 change to whisper mode, and pressing 6 will change to barge mode.
1387 * ExternalIVR now takes several options that affect the way it performs, as
1388 well as having several new commands. Please see the External IVR page on the Asterisk
1389 wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
1390 * Added ability to communicate over a TCP socket instead of forking a child process for the
1391 ExternalIVR application.
1392 * ChanIsAvail has a new option, 'a', which will return all available channels instead
1393 of just the first one if you give the function more then one channel to check.
1394 * PrivacyManager now takes an option where you can specify a context where the
1395 given number will be matched. This way you have more control over who is allowed
1396 and it stops the people who blindly enter 10 digits.
1397 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
1398 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
1399 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
1400 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
1401 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
1402 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
1403 * The Dial() application no longer copies the language used by the caller to the callee's
1404 channel. If you desire for the caller's channel's language to be used for file playback
1405 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
1406 * SendImage() no longer hangs up the channel on error; instead, it sets the
1407 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
1408 'UNSUPPORTED'. This change makes SendImage() more consistent with other
1410 * Park has a new option, 's', which silences the announcement of the parking space number.
1411 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
1412 invalid input and will be assumed to mean that no timeout is desired.
1416 * Added DNS manager support to registrations for peers referencing peer entries.
1417 DNS manager runs in the background which allows DNS lookups to be run asynchronously
1418 as well as periodically updating the IP address. These properties allow for
1419 better performance as well as recovery in the event of an IP change.
1420 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
1421 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
1422 These changes also provide performance improvements for call setup and tear down.
1423 * Added ability to specify registration expiry time on a per registration basis in
1425 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
1427 * Added t38pt_usertpsource option. See sip.conf.sample for details.
1428 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
1429 * 'sip show peers' and 'sip show users' display their entries sorted in
1430 alphabetical order, as opposed to the order they were in, in the config
1432 * Videosupport now supports an additional option, "always", which always sets
1433 up video RTP ports, even on clients that don't support it. This helps with
1434 callfiles and certain transfers to ensure that if two video phones are
1435 connected, they will always share video feeds.
1439 * Existing DNS manager lookups extended to check for SRV records.
1440 * IAX2 encryption support has been improved to support periodic key rotation
1441 within a call for enhanced security. The option "keyrotate" has been
1442 provided to disable this functionality to preserve backwards compatibility
1443 with older versions of IAX2 that do not support key rotation.
1447 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
1448 data tree based on the given <path>.
1449 * New CLI command "data show providers" that will display all the registered
1451 * New CLI command, "config reload <file.conf>" which reloads any module that
1452 references that particular configuration file. Also added "config list"
1453 which shows which configuration files are in use.
1454 * New CLI commands, "pri show version" and "ss7 show version" that will
1455 display which version of libpri and libss7 are being used, respectively.
1456 A new API call was added so trunk will now have to be compiled against
1457 a versions of libpri and libss7 that have them or it will not know that
1458 these libraries exist.
1459 * The commands "core show globals", "core set global" and "core set chanvar" has
1460 been deprecated in favor of the more semanticly correct "dialplan show globals",
1461 "dialplan set chanvar" and "dialplan set global".
1462 * New CLI command "dialplan show chanvar" to list all variables associated
1463 with a given channel.
1467 * Addresses managed by DNS manager now can check to see if there is a DNS
1468 SRV record for a given domain and will use that hostname/port if present.
1470 AMI - The manager (TCP/TLS/HTTP)
1471 --------------------------------
1472 * The Status command now takes an optional list of variables to display
1473 along with channel status.
1474 * The QueueEntry event now also includes the channel's uniqueid
1478 * res_odbc no longer has a limit of 1023 total possible unshared connections,
1479 as some people were running into this limit. This limit has been increased
1484 * The TRANSFER queue log entry now includes the the caller's original
1485 position in the transferred-from queue.
1486 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
1487 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
1488 as well as an explanation about timeout options in general
1489 * Added a new option - C - for forcing the "answered elsewhere" flag on
1490 cancellation of calls in to members of the queue. This is to avoid the
1491 call to a member of a queue having the call listed as a "missed call".
1495 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
1496 adaptive capabilities. What this means in practical terms is that if your
1497 realtime table lacks critical fields, Asterisk will now emit warnings to
1498 that effect. Also, some of the realtime drivers have the ability (if
1499 configured) to automatically add those columns to the table with the
1500 correct type and length.
1504 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
1505 the 'setvar' option to cause a given audio file to be played upon completion
1506 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
1507 Skinny channels only.
1508 * You can now compile Asterisk against the Hoard Memory Allocator, see the
1509 Hoard page on the Asterisk wiki for more information:
1510 https://wiki.asterisk.org/wiki/x/pQBB
1511 * Config file variables may now be appended to, by using the '+=' append
1512 operator. This is most helpful when working with long SQL queries in
1513 func_odbc.conf, as the queries no longer need to be specified on a single
1515 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
1516 which will add a second to the billsec when the ending
1517 time is set, if the number in the microseconds field of the end time is
1518 greater than the number of microseconds in the answer time. This allows
1519 users to count the 'initiated' seconds in their billing records.
1521 ------------------------------------------------------------------------------
1522 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
1523 ------------------------------------------------------------------------------
1525 AMI - The manager (TCP/TLS/HTTP)
1526 --------------------------------
1527 * Manager has undergone a lot of changes, all of them documented
1528 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
1529 * Manager version has changed to 1.1
1530 * Added a new action 'CoreShowChannels' to list currently defined channels
1531 and some information about them.
1532 * Added a new action 'SIPshowregistry' to list SIP registrations.
1533 * Added TLS support for the manager interface and HTTP server
1534 * Added the URI redirect option for the built-in HTTP server
1535 * The output of CallerID in Manager events is now more consistent.
1536 CallerIDNum is used for number and CallerIDName for name.
1537 * Enable https support for builtin web server.
1538 See configs/http.conf.sample for details.
1539 * Added a new action, GetConfigJSON, which can return the contents of an
1540 Asterisk configuration file in JSON format. This is intended to help
1541 improve the performance of AJAX applications using the manager interface
1543 * SIP and IAX manager events now use "ChannelType" in all cases where we
1544 indicate channel driver. Previously, we used a mixture of "Channel"
1545 and "ChannelDriver" headers.
1546 * Added a "Bridge" action which allows you to bridge any two channels that
1547 are currently active on the system.
1548 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
1549 the voicemail users setup.
1550 * Added 'DBDel' and 'DBDelTree' manager commands.
1551 * cdr_manager now reports events via the "cdr" level, separating it from
1552 the very verbose "call" level.
1553 * Manager users are now stored in memory. If you change the manager account
1554 list (delete or add accounts) you need to reload manager.
1555 * Added Masquerade manager event for when a masquerade happens between
1557 * Added "manager reload" command for the CLI
1558 * Lots of commands that only provided information are now allowed under the
1559 Reporting privilege, instead of only under Call or System.
1560 * The IAX* commands now require either System or Reporting privilege, to
1561 mirror the privileges of the SIP* commands.
1562 * Added ability to retrieve list of categories in a config file.
1563 * Added ability to retrieve the content of a particular category.
1564 * Added ability to empty a context.
1565 * Created new action to create a new file.
1566 * Updated delete action to allow deletion by line number with respect to category.
1567 * Added new action insert to add new variable to category at specified line.
1568 * Updated action newcat to allow new category to be inserted in file above another
1570 * Added new event "JitterBufStats" in the IAX2 channel
1571 * Originate now requires the Originate privilege and, if you want to call out
1572 to a subshell, it requires the System privilege, as well. This was done to
1573 enhance manager security.
1574 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
1575 * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
1576 or manager show command Atxfer from the CLI
1577 * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
1578 details or manager show command IAXregistry from the CLI
1582 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
1583 state in the dialplan, as well as creating custom device states that are
1584 controllable from the dialplan.
1585 * Extend CALLERID() function with "pres" and "ton" parameters to
1586 fetch string representation of calling number presentation indicator
1587 and numeric representation of type of calling number value.
1588 * MailboxExists converted to dialplan function
1589 * A new option to Dial() for telling IP phones not to count the call
1590 as "missed" when dial times out and cancels.
1591 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
1592 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
1593 held for any given channel. Also, locks are automatically freed when a
1595 * Added HINT() dialplan function that allows retrieving hint information.
1596 Hints are mappings between extensions and devices for the sake of
1597 determining the state of an extension. This function can retrieve the list
1598 of devices or the name associated with a hint.
1599 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
1601 * Added SYSINFO() dialplan function which allows retrieval of system information
1602 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
1603 the existence of a dialplan target.
1604 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
1605 upper and lower case, respectively.
1606 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
1607 ID for the call (not the Asterisk call ID or unique ID), provided that the
1608 channel driver supports this. For SIP, you get the SIP call-ID for the
1609 bridged channel which you can store in the CDR with a custom field.
1613 * Added CLI permissions, config file: cli_permissions.conf
1614 default is to allow all commands for every local user/group.
1615 Also this new feature added three new CLI commands:
1616 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
1617 - cli reload permissions
1618 - cli show permissions
1619 * New CLI command "core show hint" (usage: core show hint <exten>)
1620 * New CLI command "core show settings"
1621 * Added 'core show channels count' CLI command.
1622 * Added the ability to set the core debug and verbose values on a per-file basis.
1623 * Added 'queue pause member' and 'queue unpause member' CLI commands
1624 * Ability to set process limits ("ulimit") without restarting Asterisk
1625 * Enhanced "agi debug" to print the channel name as a prefix to the debug
1626 output to make debugging on busy systems much easier.
1627 * New CLI commands "dialplan set extenpatternmatching true/false"
1628 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
1629 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
1630 listed in the startup_commands section of cli.conf will get executed.
1631 * Added a CLI command, "devstate change", which allows you to set custom device
1632 states from the func_devstate module that provides the DEVICE_STATE() function
1633 and handling of the "Custom:" devices.
1634 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
1635 sorted into the different possible callbacks, with the number of entries
1636 currently scheduled for each. Gives you a feel for how busy the sip channel
1638 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
1639 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
1640 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
1644 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
1645 option is enabled, Asterisk will watch for a CNG tone in the incoming audio
1646 for a received call. If it is detected, the channel will jump to the
1647 'fax' extension in the dialplan.
1648 * The default SIP useragent= identifier now includes the Asterisk version
1649 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
1650 If set, and the incoming request carries authentication info,
1651 the username to match in the users list is taken from the Digest header
1652 rather than from the From: field. This feature is considered experimental.
1653 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
1654 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
1655 * The "localmask" setting was removed in version 1.2 and the reminder about it
1656 being removed is now also removed.
1657 * A new option "busylevel" for setting a level of calls where asterisk reports
1658 a device as busy, to separate it from call-limit. This value is also added
1659 to the SIP_PEER dialplan function.
1660 * A new realtime family called "sipregs" is now supported to store SIP registration
1661 data. If this family is defined, "sippeers" will be used for configuration and
1662 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
1663 registration data, as before.
1664 * The SIPPEER function have new options for port address, call and pickup groups
1665 * Added support for T.140 realtime text in SIP/RTP
1666 * The "checkmwi" option has been removed from sip.conf, as it is no longer
1667 required due to the restructuring of how MWI is handled. See the descriptions
1668 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
1669 for more information.
1670 * Added rtpdest option to CHANNEL() dialplan function.
1671 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
1672 * SIP now adds a header to the CANCEL if the call was answered by another phone
1673 in the same dial command, or if the new c option in dial() is used.
1674 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
1675 states it is not needed. For phones, however, that do require it the "registertrying" option
1676 has been added so it can be enabled.
1677 * A new option called "callcounter" (global/peer/user level) enables call counters needed
1678 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
1679 used to enable this functionality).
1680 * New settings for timer T1 and timer B on a global level or per device. This makes it
1681 possible to force timeout faster on non-responsive SIP servers. These settings are
1682 considered advanced, so don't use them unless you have a problem.
1683 * Added a dial string option to be able to set the To: header in an INVITE to any
1685 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
1686 the qualify frequency.
1687 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
1688 were not properly torn down due to network or endpoint failures during an established
1690 * Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB
1691 and configs/sip.conf.sample for more information on how it is used.
1692 * Added a new configuration option "authfailureevents" that enables manager events when
1693 a peer can't authenticate properly.
1694 * Added DNS manager support to registrations for peers not referencing a peer entry.
1698 * Added the trunkmaxsize configuration option to chan_iax2.
1699 * Added the srvlookup option to iax.conf
1700 * Added support for OSP. The token is set and retrieved through the CHANNEL()
1703 XMPP Google Talk/Jingle changes
1704 -------------------------------
1705 * Added the bindaddr option to gtalk.conf.
1709 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
1710 * Proper codec support in chan_skinny.
1711 * Added settings for IP and Ethernet QoS requests
1715 * Added separate settings for media QoS in mgcp.conf
1717 Console Channel Driver changes
1718 ------------------------------
1719 * Added experimental support for video send & receive to chan_oss.
1720 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
1723 Phone channel changes (chan_phone)
1724 ----------------------------------
1725 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
1727 H.323 channel Changes
1728 ---------------------
1729 * H323 remote hold notification support added (by NOTIFY message
1730 and/or H.450 supplementary service)
1732 Local channel changes
1733 ---------------------
1734 * The device state functionality in the Local channel driver has been updated
1735 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
1736 to just UNKNOWN if the extension exists.
1737 * Added jitterbuffer support for chan_local. This allows you to use the
1738 generic jitterbuffer on incoming calls going to Asterisk applications.
1739 For example, this would allow you to use a jitterbuffer for an incoming
1740 SIP call to Voicemail by putting a Local channel in the middle. This
1741 feature is enabled by using the 'j' option in the Dial string to the Local
1742 channel in conjunction with the existing 'n' option for local channels.
1743 * A 'b' option has been added which causes chan_local to return the actual channel
1744 that is behind it when queried. This is useful for transfer scenarios as the
1745 actual channel will be transferred, not the Local channel.
1747 Agent channel changes
1748 ----------------------
1749 * The ackcall and endcall options are now supplemented with options acceptdtmf
1750 and enddtmf. These allow for the DTMF keypress to be configurable. The options
1751 default to their old hard-coded values ('#' and '*' respectively) so this should
1752 not break any existing agent installations.
1754 DAHDI channel driver (chan_dahdi) Changes
1755 ----------------------------------------
1756 * SS7 support (via libss7 library)
1757 * In India, some carriers transmit CID via dtmf. Some code has been added
1758 that will handle some situations. The cidstart=polarity_IN choice has been added for
1759 those carriers that transmit CID via dtmf after a polarity change.
1760 * CID matching information is now shown when doing 'dialplan show'.
1761 * Added dahdi show version CLI command.
1762 * Added setvar support to chan_dahdi.conf channel entries.
1763 * Added two new options: mwimonitor and mwimonitornotify. These options allow
1764 you to enable MWI monitoring on FXO lines. When the MWI state changes,
1765 the script specified in the mwimonitornotify option is executed. An internal
1766 event indicating the new state of the mailbox is also generated, so that
1767 the normal MWI facilities in Asterisk work as usual.
1768 * Added signalling type 'auto', which attempts to use the same signalling type
1769 for a channel as configured in DAHDI. This is primarily designed for analog
1770 ports, but will also work for digital ports that are configured for FXS or FXO
1771 signalling types. This mode is also the default now, so if your chan_dahdi.conf
1772 does not specify signalling for a channel (which is unlikely as the sample
1773 configuration file has always recommended specifying it for every channel) then
1774 the 'auto' mode will be used for that channel if possible.
1775 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
1776 state for a channel; also ensured that the DNDState Manager event is
1777 emitted no matter how the DND state is set or cleared.
1781 * Added a new channel driver, chan_unistim. See the Asterisk wiki at
1782 https://wiki.asterisk.org/wiki/x/vgsiAQ and configs/unistim.conf.sample
1783 for details. This new channel driver allows you to use Nortel i2002,
1784 i2004, and i2050 phones with Asterisk.
1785 * Added a new channel driver, chan_console, which uses portaudio as a cross
1786 platform audio interface. It was written as a channel driver that would
1787 work with Mac CoreAudio, but portaudio supports a number of other audio
1788 interfaces, as well. Note that this channel driver requires v19 or higher
1789 of portaudio; older versions have a different API.
1793 * Added the ability to specify arguments to the Dial application when using
1794 the DUNDi switch in the dialplan.
1795 * Added the ability to set weights for responses dynamically. This can be
1796 done using a global variable or a dialplan function. Using the SHELL()
1797 function would allow you to have an external script set the weight for
1799 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
1800 functions will allow you to initiate a DUNDi query from the dialplan,
1801 find out how many results there are, and access each one.
1802 * Added the ability to specifiy a port for a dundi peer.
1806 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
1807 functions will allow you to initiate an ENUM lookup from the dialplan,
1808 and Asterisk will cache the results. ENUMRESULT can be used to access
1809 the results without doing multiple DNS queries.
1813 * Added the ability to customize which sound files are used for some of the
1814 prompts within the Voicemail application by changing them in voicemail.conf
1815 * Added the ability for the "voicemail show users" CLI command to show users
1816 configured by the dynamic realtime configuration method.
1817 * MWI (Message Waiting Indication) handling has been significantly
1818 restructured internally to Asterisk. It is now totally event based
1819 instead of polling based. The voicemail application will notify other
1820 modules that have subscribed to MWI events when something in the mailbox
1822 This also means that if any other entity outside of Asterisk is changing
1823 the contents of mailboxes, then the voicemail application still needs to
1824 poll for changes. Examples of situations that would require this option
1825 are web interfaces to voicemail or an email client in the case of using
1826 IMAP storage. So, two new options have been added to voicemail.conf
1827 to account for this: "pollmailboxes" and "pollfreq". See the sample
1828 configuration file for details.
1829 * Added "tw" language support
1830 * Added support for storage of greetings using an IMAP server
1831 * Added ability to customize forward, reverse, stop, and pause keys for message playback
1832 * SMDI is now enabled in voicemail using the smdienable option.
1833 * A "lockmode" option has been added to asterisk.conf to configure the file
1834 locking method used for voicemail, and potentially other things in the
1835 future. The default is the old behavior, lockfile. However, there is a
1836 new method, "flock", that uses a different method for situations where the
1837 lockfile will not work, such as on SMB/CIFS mounts.
1838 * Added the ability to backup deleted messages, to ease recovery in the case
1839 that a user accidentally deletes a message, and discovers that they need it.
1840 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
1841 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
1842 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
1843 voicemail boxes. The SMDI interface can also poll for MWI changes when some
1844 outside entity is modifying the state of the mailbox (such as IMAP storage or
1845 a web interface of some kind).
1846 * Added the support for marking messages as "urgent." There are two methods to accomplish
1847 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
1848 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
1849 the message as urgent after he has recorded a voicemail by following the voice instructions.
1850 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
1855 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
1856 used across multiple queues.
1857 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
1858 setqueueentryvar options for each queue, see queues.conf.sample for details.
1859 * Added keepstats option to queues.conf which will keep queue
1860 statistics during a reload.
1861 * setinterfacevar option in queues.conf also now sets a variable
1862 called MEMBERNAME which contains the member's name.
1863 * Added 'Strategy' field to manager event QueueParams which represents
1864 the queue strategy in use.
1865 * Added option to run macro when a queue member is connected to a caller,
1866 see queues.conf.sample for details.
1867 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
1868 does not count paused queue members as unavailable.
1869 * Added min-announce-frequency option to queues.conf which allows you to control the
1870 minimum amount of time between queue announcements for use when the caller's queue
1871 position changes frequently.
1872 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
1874 * Added ability for non-realtime queues to have realtime members
1875 * Added the "linear" strategy to queues.
1876 * Added the "wrandom" strategy to queues.
1877 * Added new channel variable QUEUE_MIN_PENALTY
1878 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
1879 rules in queuerules.conf. See configs/queuerules.conf.sample for details
1880 * Added a new parameter for member definition, called state_interface. This may be
1881 used so that a member may be called via one interface but have a different interface's
1882 device state reported.
1883 * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
1884 "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
1885 "manager show command QueueReset."
1886 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
1887 specified by the periodic-announce option, then one will be chosen randomly when it is time
1888 to play a periodic announcment
1889 * New configuration options: announce-position now takes two more values in addition to "yes" and
1890 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
1891 announce-position-limit. By setting announce-position to "limit" callers will only have their
1892 position announced if their position is less than what is specified by announce-position-limit.
1893 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
1894 will be told that their are more than announce-position-limit callers waiting.
1895 * Two new queue log events have been added. An ADDMEMBER event will be logged
1896 when a realtime queue member is added and a REMOVEMEMBER event will be logged
1897 when a realtime queue member is removed. Since there is no calling channel associated
1898 with these events, the string "REALTIME" is placed where the channel's unique id
1899 is typically placed.
1900 * The configuration method for the "joinempty" and "leavewhenempty" options has
1901 changed to a comma-separated list of methods of determining member availability
1902 instead of vague terms such as "yes," "loose," "no," and "strict." These old four
1903 values are still accepted for backwards-compatibility, though.
1904 * The average talktime is now calculated on queues. This information is reported via the
1905 CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
1906 and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
1911 * The 'o' option to provide an optimization has been removed and its functionality
1912 has been enabled by default.
1913 * When a conference is created, the UNIQUEID of the channel that caused it to be
1914 created is stored. Then, every channel that joins the conference will have the
1915 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
1916 callers that come and go from long standing conferences.
1917 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
1918 except it does operations on a channel by name, instead of number in a conference.
1919 This is a very useful feature in combination with the 'X' option to ChanSpy.
1920 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
1922 * Added new RealTime functionality to provide support for scheduled conferencing.
1923 This includes optional messages to the caller if they attempt to join before
1924 the schedule start time, or to allow the caller to join the conference early.
1925 Also included is optional support for limiting the number of callers per
1926 RealTime conference.
1927 * Added the S() and L() options to the MeetMe application. These are pretty
1928 much identical to the S() and L() options to Dial(). They let you set
1929 timeouts for the conference, as well as have warning sounds played to
1930 let the caller know how much time is left, and when it is running out.
1931 * Added the ability to do "meetme concise" with the "meetme" CLI command.
1932 This extends the concise capabilities of this CLI command to include
1933 listing all conferences, instead of an addition to the other sub commands
1934 for the "meetme" command.
1935 * Added the ability to specify the music on hold class used to play into the
1936 conference when there is only one member and the M option is used.
1937 * Added MEETME_INFO dialplan function which provides a way to query
1938 various properties of a Meetme conference.
1939 * Added new admin features: *81: Roll call, *82: eject all, *83: mute all,
1940 and *84: record in-conf
1942 Other Dialplan Application Changes
1943 ----------------------------------
1944 * Argument support for Gosub application
1945 * From the to-do lists: straighten out the app timeout args:
1946 Wait() app now really does 0.3 seconds- was truncating arg to an int.
1947 WaitExten() same as Wait().
1948 Congestion() - Now takes floating pt. argument.
1949 Busy() - now takes floating pt. argument.
1950 Read() - timeout now can be floating pt.
1951 WaitForRing() now takes floating pt timeout arg.
1952 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
1953 * Added 's' option to Page application.
1954 * Added an optional timeout argument to the Page application.
1955 * Added 'E', 'V', and 'P' commands to ExternalIVR.
1956 * Added 'o' and 'X' options to Chanspy.
1957 * Added a new dialplan application, Bridge, which allows you to bridge the
1958 calling channel to any other active channel on the system.
1959 * Added the ability to specify a music on hold class to play instead of ringing
1960 for the SLATrunk application.
1961 * The Read application no longer exits the dialplan on error. Instead, it sets
1962 READSTATUS to ERROR, which you can catch and handle separately.
1963 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
1964 of asking for verification of each name, one at a time.
1965 * Privacy() no longer uses privacy.conf, as all options are specifyable as
1966 direct options to the app.
1967 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
1969 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
1970 * The ChannelRedirect application no longer exits the dialplan if the given channel
1971 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
1972 or NOCHANNEL if the given channel was not found.
1973 * The silencethreshold setting that was previously configurable in multiple
1974 applications is now settable globally via dsp.conf.
1976 Music On Hold Changes
1977 ---------------------
1978 * A new option, "digit", has been added for music on hold classes in
1979 musiconhold.conf. If this is set for a music on hold class, a caller
1980 listening to music on hold can press this digit to switch to listening
1981 to this music on hold class.
1982 * Support for realtime music on hold has been added.
1983 * In conjunction with the realtime music on hold, a general section has
1984 been added to musiconhold.conf, its sole variable is cachertclasses. If this
1985 is set, then music on hold classes found in realtime will be cached in memory.
1989 * AEL upgraded to use the Gosub with Arguments instead
1990 of Macro application, to hopefully reduce the problems
1991 seen with the artificially low stack ceiling that
1992 Macro bumps into. Macros can only call other Macros
1993 to a depth of 7. Tests run using gosub, show depths
1994 limited only by virtual memory. A small test demonstrated
1995 recursive call depths of 100,000 without problems.
1996 -- in addition to this, all apps that allowed a macro
1997 to be called, as in Dial, queues, etc, are now allowing
1998 a gosub call in similar fashion.
1999 * AEL now generates LOCAL(argname) declarations when it
2000 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
2001 etc. That makes the arguments local in scope. The user
2002 can define their own local variables in macros, now,
2003 by saying "local myvar=someval;" or using Set() in this
2004 fashion: Set(LOCAL(myvar)=someval); ("local" is now
2006 * utils/conf2ael introduced. Will convert an extensions.conf
2007 file into extensions.ael. Very crude and unfinished, but
2008 will be improved as time goes by. Should be useful for a
2009 first pass at conversion.
2010 * aelparse will now read extensions.conf to see if a referenced
2011 macro or context is there before issueing a warning.
2012 * AEL parser sets a local channel variable ~~EXTEN~~, to
2013 preserve the value of ${EXTEN} thru switch statements.
2014 * New operator in $[...] expressions: the ~~ operator serves
2015 as a concatenation operator. AT THE MOMENT, it is really only
2016 necessary and useful in AEL, especially in if() expressions.
2017 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
2018 any enclosing double-quotes, and evaluate to the value of a
2019 concatenated with the value of b. For example if a is set to
2020 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
2021 evaluate to xyzabc .
2024 Call Features (res_features) Changes
2025 ------------------------------------
2026 * Added the parkedcalltransfers option to features.conf
2027 * Added parkedcallparking option to control one touch parking w/ parking
2029 * Added parkedcallhangup option to control disconnect feature w/ parking
2031 * Added parkedcallrecording option to control one-touch record w/ parking
2033 * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
2034 parkedcalltransfers option support for multiple parking lots.
2035 * Added BRIDGE_FEATURES variable to set available features for a channel
2036 * The built-in method for doing attended transfers has been updated to
2037 include some new options that allow you to have the transferee sent
2038 back to the person that did the transfer if the transfer is not successful.
2039 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
2040 in features.conf.sample.
2041 * Added support for configuring named groups of custom call features in
2042 features.conf. This means that features can be written a single time, and
2043 then mapped into groups of features for different key mappings or easier
2045 * Updated the ParkedCall application to allow you to not specify a parking
2046 extension. If you don't specify a parking space to pick up, it will grab
2047 the first one available.
2048 * Added cli command 'features reload' to reload call features from features.conf
2049 * Moved into core asterisk binary.
2050 * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
2051 * Added the ability for custom parking lots to be configured with their own
2052 parking extension with the parkext option.
2054 Language Support Changes
2055 ------------------------
2056 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
2057 * Added support for the Hungarian language for saying numbers, dates, and times.
2061 * Added SPEECH commands for speech recognition. A complete listing can be found
2063 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
2064 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
2065 does not behave as expected; the native command needs to be used, instead.
2066 * Added the ability to perform SRV lookups on fast AGI calls. To use this
2067 feature, simply use hagi: instead of agi: as the protocol portion
2068 of the URI parameter to the AGI function call in your dial plan. Also note
2069 that specifying a port number in the AGI URI will disable SRV lookups,
2070 even if you use the hagi: protocol.
2071 * No longer support MSG_OOB flag on HANGUP.
2075 * Added rotatestrategy option to logger.conf, along with two new options:
2076 "timestamp" which will use the time to name the logger files instead of
2077 sequence number; and "rotate", which rotates the names of the log files,
2078 similar to the way syslog rotates files.
2079 * Added exec_after_rotate option to logger.conf, which allows a system
2080 command to be run after rotation. This is primarily useful with
2081 rotatestrategy=rotate, to allow a limit on the number of log files kept
2082 and to ensure that the oldest log file gets deleted.
2083 * Added realtime support for the queue log
2087 * The cdr_manager module has a [mappings] feature, like cdr_custom,
2088 to add fields to the manager event from the CDR variables.
2089 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
2090 backend database CDR table. Specifically, additional, non-standard
2091 columns are supported, merely by setting the corresponding CDR variable in
2092 your dialplan. In addition, you may alias any column to another name (for
2093 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
2094 simply "alias src => ANI" in the configuration file). Records may be
2095 posted to more than one backend, simply by specifying multiple categories
2096 in the configuration file. And finally, you may filter which CDRs get
2097 posted to each backend, by specifying a filter (which the record must
2098 match) for the particular category. Filters are additive (meaning all
2099 rules must match to post that CDR).
2100 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
2101 module. Specifically, you may add additional columns into the table and
2102 they will be set, if you set the corresponding CDR variable name. Also,
2103 if you omit columns in your database table, they will be silently skipped
2104 (but a record will still be inserted, based on what columns remain). Note
2105 that the other two features from cdr_adaptive_odbc (alias and filter) are
2106 not currently supported.
2107 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
2108 has been disabled using the NoCDR application.
2110 Miscellaneous New Modules
2111 -------------------------
2112 * Added a new CDR module, cdr_sqlite3_custom.
2113 * Added a new realtime configuration module, res_config_sqlite
2114 * Added a new codec translation module, codec_resample, which re-samples
2115 signed linear audio between 8 kHz and 16 kHz to help support wideband
2117 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
2118 based on configuration templates that use Asterisk dialplan function and
2119 variable substitution. It should be possible to create phone profiles and
2120 templates that work for the majority of phones provisioned over http. It
2121 is currently only intended to provision a single user account per phone.
2122 An example profile and set of templates for Polycom phones is provided.
2123 NOTE: Polycom firmware is not included, but should be placed in
2124 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
2125 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
2126 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
2127 provided; there is a JACK() application, and a JACK_HOOK() function. Both
2128 interfaces create an input and output JACK port. The application makes
2129 these ports the endpoint of the call. The audio coming from the channel
2130 goes out the output port and whatever comes back in on the input port is
2131 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
2132 audiohook on the channel. This lets you run the audio coming from a
2133 channel through JACK, and whatever comes back in is what gets forwarded
2134 on as the channel's audio. This is very useful for building custom
2135 vocoders or doing recording or analysis of the channel's audio in another
2137 * Added a new module, res_config_curl, which permits using a HTTP POST url
2138 to retrieve, create, update, and delete realtime information from a remote
2139 web server. Note that this module requires func_curl.so to be loaded for
2140 backend functionality.
2141 * Added a new module, res_config_ldap, which permits the use of an LDAP
2142 server for realtime data access.
2143 * Added support for writing and running your dialplan in lua using the pbx_lua
2144 module. See configs/extensions.lua.sample for examples of how to do this.
2148 * Ability to use libcap to set high ToS bits when non-root
2149 on Linux. If configure is unable to find libcap then you
2150 can use --with-cap to specify the path.
2151 * Added maxfiles option to options section of asterisk.conf which allows you to specify
2152 what Asterisk should set as the maximum number of open files when it loads.
2153 * Added the jittertargetextra configuration option.
2154 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
2155 configuration files for the IP channel drivers. The new option is "cos".
2156 This information is also documented on the Asterisk wiki at
2157 https://wiki.asterisk.org/wiki/x/EYBG
2158 * When originating a call using AMI or pbx_spool that fails the reason for failure
2159 will now be available in the failed extension using the REASON dialplan variable.
2160 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
2161 It allows you to configure a prefix for auto-monitor recordings.
2162 * A new extension pattern matching algorithm, based on a trie, is introduced
2163 here, that could noticeably speed up mid-sized to large dialplans.
2164 It is NOT used by default, as duplicating the behaviour of the old pattern
2165 matcher is still under development. A config file option, in extensions.conf,
2166 in the [general] section, called "extenpatternmatchingnew", is by default
2167 set to false; setting that to true will force the use of the new algorithm.
2168 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
2169 be used to switch the algorithms at run time.
2170 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
2171 specifying which socket to use to connect to the running Asterisk daemon
2173 * Performance enhancements to the sched facility, which is used in
2174 the channel drivers, etc. Added hashtabs and doubly-linked lists
2175 to speed up deletion; start at the beginning or end of list to
2177 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
2178 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
2179 Added regression tests to the tests/ dir, also.
2180 * Added a refcount trace feature to astobj2 for those trying to balance
2181 object creation, deletion; work, play; space and time. See the
2182 notes in astobj2.h. Also, see utils/refcounter as well, as a
2183 quick way to find unbalanced refcounts in what could be a sea
2184 of objects that were balanced.
2185 * Added logging to 'make update' command. See update.log
2186 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
2187 do not come from the remote party.
2188 * Added the 'n' option to the SpeechBackground application to tell it to not
2189 answer the channel if it has not already been answered.
2190 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
2191 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
2193 * iLBC source code no longer included (see UPGRADE.txt for details)
2194 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
2195 deadlock is detected, a backtrace of the stack which led to the lock calls
2196 will be output to the CLI.
2197 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
2198 the "core show locks" CLI command will give lock information output as well
2199 as a backtrace of the stack which led to the lock calls.
2200 * users.conf now sports an optional alternateexts property, which permits
2201 allocation of additional extensions which will reach the specified user.
2202 * A new option for the configure script, --enable-internal-poll, has been added
2203 for use with systems which may have a buggy implementation of the poll system
2204 call. If you notice odd behavior such as the CLI being unresponsive on remote
2205 consoles, you may want to try using this option. This option is enabled by default
2206 on Darwin systems since it is known that the Darwin poll() implementation has
2210 --------------------
2211 * In addition to timing from DAHDI, there is a new timing module called
2212 res_timing_timerfd. In order to use this, you must be running Linux with
2213 a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
2214 script will be able to tell if you have the requirements. From menuselect, select
2215 res_timing_timerfd from the Resource Modules menu.
projects / asterisk/asterisk.git / blob