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 ==============================================================================
10 ------------------------------------------------------------------------------
11 --- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
12 ------------------------------------------------------------------------------
15 AMI (Asterisk Manager Interface)
17 * The SIPshowpeer action will now include a 'SubscribeContext' field for a peer
18 in its response if the peer has a subscribe context set.
20 * The SIPqualifypeer action now acknowledges the request once it has established
21 that the request is against a known peer. It also issues a new event,
22 'SIPqualifypeerdone', once the qualify action has been completed.
24 * The PlayDTMF action now supports an optional 'Duration' parameter. This
25 specifies the duration of the digit to be played, in milliseconds.
27 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
28 updates when changes occur instead of requiring the use of pollmailboxes.
30 * CLI Command 'Manager Show Commands' no longer truncates command names longer
31 than 15 characters and no longer shows authorization requirement for commands.
32 'Manager Show Command' now displays the privileges needed for using a given
33 manager command instead.
35 * Added new action "ControlPlayback". The ControlPlayback action allows an AMI
36 client to manipulate audio currently being played back on a channel. The
37 supported operations depend on the application being used to send audio to
38 the channel. When the audio playback was initiated using the ControlPlayback
39 application or CONTROL STREAM FILE AGI command, the audio can be paused,
40 stopped, restarted, reversed, or skipped forward. When initiated by other
41 mechanisms (such as the Playback application), the audio can be stopped,
42 reversed, or skipped forward.
49 * Added general support for busy detection.
51 * Added ECAM command support for Sony Ericsson phones.
56 * The BRIDGE_FEATURES channel variable would previously only set features for
57 the calling party and would set this feature regardless of whether the
58 feature was in caps or in lowercase. Use of a caps feature for a letter
59 will now apply the feature to the calling party while use of a lowercase
60 letter will apply that feature to the called party.
62 * Add support for automixmonitor to the BRIDGE_FEATURES channel variable.
66 * When performing queue pause/unpause on an interface without specifying an
67 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
68 least one member of any queue exists for that interface.
70 * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
71 for realtime queue log entries.
75 * Add queue available hint. exten => 8501,hint,Queue:markq_avail
76 Note: the suffix '_avail' after the queuename.
77 Reports 'InUse' for no logged in agents or no free agents.
78 Reports 'Idle' when an agent is free.
82 * Redirecting reasons can now be set to arbitrary strings. This means
83 that the REDIRECTING dialplan function can be used to set the redirecting
84 reason to any string. It also allows for custom strings to be read as the
85 redirecting reason from SIP Diversion headers.
87 ------------------------------------------------------------------------------
88 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
89 ------------------------------------------------------------------------------
93 * The Asterisk build system will now build and install a shared library
94 (libasteriskssl.so) used to wrap various initialization and shutdown functions
95 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
96 that Asterisk can ensure that these functions do *not* get called by any
97 modules that are loaded into Asterisk, since they should only be called once
98 in any single process. If desired, this feature can be disabled by supplying
99 the "--disable-asteriskssl" option to the configure script.
101 * A new make target, 'full', has been added to the Makefile. This performs
102 the same compilation actions as make all, but will also scan the entirety of
103 each source file for documentation. This option is needed to generate AMI
104 event documentation. Note that your system must have Python in order for
105 this make target to succeed.
107 * The optimization portion of the build system has been reworked to avoid
108 broken builds on certain architectures. All architecture-specific
109 optimization has been removed in favor of using -march=native to allow gcc
110 to detect the environment in which it is running when possible. This can
111 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
113 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
114 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
116 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
117 previously parsed the header file to obtain the version of Asterisk, you
118 will now have to go through Asterisk to get the version information.
126 * Added 'F()' option. Similar to the dial option, this can be supplied with
127 arguments indicating where the callee should go after the caller is hung up,
128 or without options specified, the priority after the Queue will be used.
133 * Added menu action admin_toggle_mute_participants. This will mute / unmute
134 all non-admin participants on a conference. The confbridge configuration
135 file also allows for the default sounds played to all conference users when
136 this occurs to be overriden using sound_participants_unmuted and
137 sound_participants_muted.
139 * Added menu action participant_count. This will playback the number of
140 current participants in a conference.
142 * Added announcement configuration option to user profile. If set the sound
143 file will be played to the user, and only the user, upon joining the
149 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
150 channels respectively before the callee channels are called.
155 * Added support for IPv6.
157 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
158 external process will cause the current playlist to be cleared, including
159 stopping any audio file that is currently playing. This is useful when you
160 want to interrupt audio playback only when specific DTMF is entered by the
166 * A new option, 'I' has been added to app_followme. By setting this option,
167 Asterisk will not update the caller with connected line changes when they
168 occur. This is similar to app_dial and app_queue.
170 * The 'N' option is now ignored if the call is already answered.
172 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
173 and caller channels respectively before the callee channels are called.
175 * The winning FollowMe outgoing call is now put on hold if the caller put it on
181 * MixMonitor hooks now have IDs associated with them which can be used to
182 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
183 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
184 now accepts that ID as an argument.
186 * Added 'm' option, which stores a copy of the recording as a voicemail in the
192 * The connect action in app_mysql now allows you to specify a port number to
193 connect to. This is useful if you run a MySQL server on a non-standard
199 * Increased the default number of allowed destinations from 5 to 12.
204 * The app_page application now no longer depends on DAHDI or app_meetme. It
205 has been re-architected to use app_confbridge internally.
210 * Added queue options autopausebusy and autopauseunavail for automatically
211 pausing a queue member when their device reports busy or congestion.
213 * The 'ignorebusy' option for queue members has been deprecated in favor of
214 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
215 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
216 per interface basis. Individual ringinuse values can now be set in
217 queues.conf via an argument to member definitions. Lastly, the queue
218 'ringinuse' setting now only determines defaults for the per member
219 'ringinuse' setting and does not override per member settings like it does
222 * Added 'F()' option. Similar to the dial option, this can be supplied with
223 arguments indicating where the callee should go after the caller is hung up,
224 or without options specified, the priority after the Queue will be used.
226 * Added new option log_member_name_as_agent, which will cause the membername to
227 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
228 state_interface has been set.
230 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
234 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
235 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
236 changed arguments to SayUnixTime so that every option is truly optional even
237 when using multiple options (so that j option could be used without having to
238 manually specify timezone and format) There are other benefits, e.g., format
239 can now be used without specifying time zone as well.
244 * Addition of the VM_INFO function - see Function changes.
246 * The imapserver, imapport, and imapflags configuration options can now be
247 overriden on a user by user basis.
249 * When voicemail plays a message's envelope with saycid set to yes, when
250 reaching the caller id field it will play a recording of a file with the same
251 base name as the sender's callerid if there is a similarly named file in
252 <astspooldir>/recordings/callerids/
254 * Voicemails now contains a unique message identifier "msg_id", which is stored
255 in the message envelope with the sound files. IMAP backends will now store
256 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
257 backends will store the message identifier in a "msg_id" column. See
258 UPGRADE.txt for more information.
260 * Added VoiceMailPlayMsg application. This application will play a single
261 voicemail message from a mailbox. The result of the application, SUCCESS or
262 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
267 * Hangup handlers can be attached to channels using the CHANNEL() function.
268 Hangup handlers will run when the channel is hung up similar to the h
269 extension. The hangup_handler_push option will push a GoSub compatible
270 location in the dialplan onto the channel's hangup handler stack. The
271 hangup_handler_pop option will remove the last added location, and optionally
272 replace it with a new GoSub compatible location. The hangup_handler_wipe
273 option will remove all locations on the stack, and optionally add a new
276 * The expression parser now recognizes the ABS() absolute value function,
277 which will convert negative floating point values to positive values.
279 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
280 control of faxdetect.
282 * Addition of the VM_INFO function that can be used to retrieve voicemail
283 user information, such as the email address and full name.
284 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
287 * The REDIRECTING function now supports the redirecting original party id
290 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
291 lets you set some of the configuration options from the [general] section
292 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
293 the key sequence used to activate built-in features, such as blindxfer,
294 and automon. See the built-in documentation for details.
296 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
297 instead of simply the uri. This is the format that MessageSend() can use
298 in the from parameter for outgoing SIP messages.
300 * Added the PRESENCE_STATE function. This allows retrieving presence state
301 information from any presence state provider. It also allows setting
302 presence state information from a CustomPresence presence state provider.
303 See AMI/CLI changes for related commands.
305 * Added the AMI_CLIENT function to make manager account attributes available
306 to the dialplan. It currently supports returning the current number of
307 active sessions for a given account.
309 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
310 and the REDIRECTING functions.
318 * Added a manager event "LocalBridge" for local channel call bridges between
319 the two pseudo-channels created.
324 * Added dialtone_detect option for analog ports to disconnect incoming
325 calls when dialtone is detected.
327 * Added option colp_send to send ISDN connected line information. Allowed
328 settings are block, to not send any connected line information; connect, to
329 send connected line information on initial connect; and update, to send
330 information on any update during a call. Default is update.
332 * Add options namedcallgroup and namedpickupgroup to support installations
333 where a higher number of groups (>64) is required.
335 * Added support to use private party ID information with PRI calls.
340 * A new channel driver named chan_motif has been added which provides support for
341 Google Talk and Jingle in a single channel driver. This new channel driver includes
342 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
343 hold, unhold, and ringing notification. It is also compliant with the current Jingle
344 specification, current Google Jingle specification, and the original Google Talk
350 * Added NAT support for RTP. Setting in config is 'nat', which can be set
351 globally and overriden on a peer by peer basis.
353 * Direct media functionality has been added. Options in config are:
354 directmedia (directrtp) and directrtpsetup (earlydirect)
356 * ChannelUpdate events now contain a CallRef header.
361 * Asterisk will no longer substitute CID number for CID name in the display
362 name field if CID number exists without a CID name. This change improves
363 compatibility with certain device features such as Avaya IP500's directory
366 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
367 created using that setting to not be removed during SIP reload.
369 * Added settings recordonfeature and recordofffeature. When receiving an INFO
370 request with a "Record:" header, this will turn the requested feature on/off.
371 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
372 dynamic features must be enabled and configured properly on the requesting
373 channel for this to function properly.
375 * Add support to realtime for the 'callbackextension' option.
377 * When multiple peers exist with the same address, but differing
378 callbackextension options, incoming requests that are matched by address
379 will be matched to the peer with the matching callbackextension if it is
382 * Two new NAT options, auto_force_rport and auto_comedia, have been added
383 which set the force_rport and comedia options automatically if Asterisk
384 detects that an incoming SIP request crossed a NAT after being sent by
387 * NAT settings are now a combinable list of options. The equivalent of the
388 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
390 * Adds an option send_diversion which can be disabled to prevent
391 diversion headers from automatically being added to INVITE requests.
393 * Add support for lightweight NAT keepalive. If enabled a blank packet will
394 be sent to the remote host at a given interval to keep the NAT mapping open.
395 This can be enabled using the keepalive configuration option.
397 * Add option 'tonezone' to specify country code for indications. This option
398 can be set both globally and overridden for specific peers.
400 * The SIP Security Events Framework now supports IPv6.
402 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
403 between multiple user agents. When set, for directmedia reinvites,
404 Asterisk will not send an immediate reinvite on an incoming call leg. This
405 option is useful when peered with another SIP user agent that is known to
406 send immediate direct media reinvites upon call establishment.
408 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
411 * Add options subminexpiry and submaxexpiry to set limits of subscription
412 timer independently from registration timer settings. The setting of the
413 registration timer limits still is done by options minexpiry, maxexpiry
414 and defaultexpiry. For backwards compatibility the setting of minexpiry
415 and maxexpiry also is used to configure the subscription timer limits if
416 subminexpiry and submaxexpiry are not set in sip.conf.
418 * Set registration timer limits to default values when reloading sip
419 configuration and values are not set by configuration.
421 * Add options namedcallgroup and namedpickupgroup to support installations
422 where a higher number of groups (>64) is required.
424 * When a MESSAGE request is received, the address the request was received from
425 is now saved in the SIP_RECVADDR variable.
427 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
428 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
429 the ANI2/OLI information is set on the channel, which can be retrieved using
430 the CALLERID function.
432 * Peers can now be configured to support negotiation of ICE candidates using
433 the setting icesupport. See res_rtp_asterisk changes for more information.
435 * Added support for format attribute negotiation. See the Codecs changes for
438 * Extra headers specified with SIPAddHeader are sent with the REFER message
439 when using Transfer application. See refer_addheaders in sip.conf.sample.
441 * Added support to use private party ID information with calls.
446 * Added skinny version 17 protocol support.
451 * Added ability to use multiple lines for a single phone. This allows multiple
452 calls to occur on a single phone, using callwaiting and switching between calls.
454 * Added option 'sharpdial' allowing end dialing by pressing # key
456 * Added option 'interdigit_timer' to control phone dial timeout
458 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
460 * Added global 'debug' option, that enables debug in channel driver
462 * Added ability to translate on-screen menu in multiple languages. Tested on
463 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
464 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
467 * In addition to English added French and Russian languages for on-screen menus
469 * Reworked dialing number input: added dialing by timeout, immediate dial on
470 on dialplan compare, phone number length now not limited by screen size
472 * Added ability to pickup a call using features.conf defined value and
478 * Add options namedcallgroup and namedpickupgroup to support installations
479 where a higher number of groups (>64) is required.
481 * Added support to use private party ID information with calls.
486 * The minimum DTMF duration can now be configured in asterisk.conf
487 as "mindtmfduration". The default value is (as before) set to 80 ms.
488 (previously it was only available in source code)
490 * Named ACLs can now be specified in acl.conf and used in configurations that
491 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
492 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
493 working ACL. In addition, some CLI commands have been added to provide
494 show information and allow for module reloading - see CLI Changes.
496 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
497 items (separated by commas), and items in the rule can be negated by prefixing
498 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
499 longer necessray to control the order that the 'permit' and 'deny' columns are
500 returned from queries.
502 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
503 be used within the dynamic weight attribute when specifying a mapping.
505 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
506 header, instead of putting the user defined event name there. When enabled
507 the UserDefType header is added for user defined events. This feature is
508 enabled with the setting show_user_defined.
510 * Macro has been deprecated in favor of GoSub. For redirecting and connected
511 line purposes use the following variables instead of their macro equivalents:
512 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
513 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
514 cc_callback_macro in channel configurations.
516 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
519 * Call files now support the "early_media" option to connect with an outgoing
520 extension when early media is received.
522 * Added support to use private party ID information with calls.
527 * A new channel variable, AGIEXITONHANGUP, has been added which allows
528 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
529 AGI application would exit immediately after a channel hangup is detected.
531 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
532 are resolved and each address is attempted in turn until one succeeds or
536 AMI (Asterisk Manager Interface)
538 * The originate action now has an option "EarlyMedia" that enables the
539 call to bridge when we get early media in the call. Previously,
540 early media was disregarded always when originating calls using AMI.
542 * Added setvar= option to manager accounts (much like sip.conf)
544 * Originate now generates an error response if the extension given is not found
547 * MixMonitor will now show IDs associated with the mixmonitor upon creating
548 them if the i(variable) option is used. StopMixMonitor will accept
549 MixMonitorID as an option to close specific MixMonitors.
551 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
552 updated to include information about peers configured with
553 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
554 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
555 returned if auto_force_rport is not enabled.
557 * Added SIPpeerstatus manager command which will generate PeerStatus events
558 similar to the existing PeerStatus events found in chan_sip on demand.
560 * Hangup now can take a regular expression as the Channel option. If you want
561 to hangup multiple channels, use /regex/ as the Channel option. Existing
562 behavior to hanging up a single channel is unchanged, but if you pass a regex,
563 the manager will send you a list of channels back that were hung up.
565 * Support for IPv6 addresses has been added.
567 * AMI Events can now be documented in the Asterisk source. Note that AMI event
568 documentation is only generated when Asterisk is compiled using 'make full'.
569 See the CLI section for commands to display AMI event information.
571 * The AMI Hangup event now includes the AccountCode header so you can easily
572 correlate with AMI Newchannel events.
574 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
575 the StateInterface of the queue member.
577 * Added AMI event SessionTimeout in the Call category that is issued when a
578 call is terminated due to either RTP stream inactivity or SIP session timer
581 * CEL events can now contain a user defined header UserDefType. See core
582 changes for more information.
584 * OOH323 ChannelUpdate events now contain a CallRef header.
586 * Added PresenceState command. This command will report the presence state for
587 the given presence provider.
589 * Added Parkinglots command. This will list all parking lots as a series of
590 AMI Parkinglot events.
592 * Added MessageSend command. This behaves in the same manner as the
593 MessageSend application, and is a technolgoy agnostic mechanism to send out
594 of call text messages.
596 * Added "message" class authorization. This grants an account permission to
597 send out of call messages. Write-only.
602 * The "dialplan add include" command has been modified to create context a context
603 if one does not already exist. For instance, "dialplan add include foo into bar"
604 will create context "bar" if it does not already exist.
606 * A "dialplan remove context" command has been added to remove a context from
609 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
610 filenames of all running mixmonitors on a channel.
612 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
613 numeric instead of 0, 1, or 2.
615 * "stun show status" will show a table describing how the STUN client is
618 * "acl show [named acl]" will show information regarding a Named ACL. The
619 acl module can be reloaded with "reload acl".
621 * Added CLI command to display AMI event information - "manager show events",
622 which shows a list of all known and documented AMI events, and "manager show
623 event [event name]", which shows detail information about a specific AMI
626 * The result of the CLI command "queue show" now includes the state interface
627 information of the queue member.
629 * The command "core set verbose" will now set a separate level of logging for
630 each remote console without affecting any other console.
632 * Added command "cdr show pgsql status" to check connection status
634 * "sip show channel" will now display the complete route set.
636 * Added "presencestate list" command. This command will list all custom
637 presence states that have been set by using the PRESENCE_STATE dialplan
640 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
641 command. This changes a custom presence to a new state.
646 * Codec lists may now be modified by the '!' character, to allow succinct
647 specification of a list of codecs allowed and disallowed, without the
648 requirement to use two different keywords. For example, to specify all
649 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
651 * Add support for parsing SDP attributes, generating SDP attributes, and
652 passing it through. This support includes codecs such as H.263, H.264, SILK,
653 and CELT. You are able to set up a call and have attribute information pass.
654 This should help considerably with video calls.
656 * The iLBC codec can now use a system-provided iLBC library if one is installed,
657 just like the GSM codec.
661 * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
662 'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
666 * Asterisk version and build information is now logged at the beginning of a
669 * Threads belonging to a particular call are now linked with callids which get
670 added to any log messages produced by those threads. Log messages can now be
671 easily identified as involved with a certain call by looking at their call id.
672 Call ids may also be attached to log messages for just about any case where
673 it can be determined to be related to a particular call.
675 * Each logging destination and console now have an independent notion of the
676 current verbosity level. Logger.conf now allows an optional argument to
677 the 'verbose' specifier, indicating the level of verbosity sent to that
678 particular logging destination. Additionally, remote consoles now each
679 have their own verbosity level. The command 'core set verbose' will now set
680 a separate level for each remote console without affecting any other
686 * Added 'announcement' option which will play at the start of MOH and between
687 songs in modes of MOH that can detect transitions between songs (eg.
693 * New per parking lot options: comebackcontext and comebackdialtime. See
694 configs/features.conf.sample for more details.
696 * Channel variable PARKER is now set when comebacktoorigin is disabled in
699 * Channel variable PARKEDCALL is now set with the name of the parking lot
700 when a timeout occurs.
706 CDR Postgresql Driver
708 * Added command "cdr show pgsql status" to check connection status
711 CDR Adaptive ODBC Driver
713 * Added schema option for databases that support specifying a schema.
721 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
722 CALENDAR_WRITE has completed successfully.
727 * A new option, 'probation' has been added to rtp.conf
728 RTP in strictrtp mode can now require more than 1 packet to exit learning
729 mode with a new source (and by default requires 4). The probation option
730 allows the user to change the required number of packets in sequence to any
731 desired value. Use a value of 1 to essentially restore the old behavior.
732 Also, with strictrtp on, Asterisk will now drop all packets until learning
733 mode has successfully exited. These changes are based on how pjmedia handles
734 media sources and source changes.
736 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
737 enabled or disabled using the icesupport setting. A variety of other
738 settings have been introduced to configure STUN/TURN connections.
743 * A new module, res_corosync, has been introduced. This module uses the
744 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
745 of Asterisk servers to both Message Waiting Indication (MWI) and/or
746 Device State (presence) information. This module is very similar to, and
747 is a replacement for the res_ais module that was in previous releases of
753 * This module adds a cleaned up, drop-in replacement for res_jabber called
754 res_xmpp. This provides the same externally facing functionality but is
755 implemented differently internally. res_jabber has been deprecated in favor
756 of res_xmpp; please see the UPGRADE.txt file for more information.
761 * The safe_asterisk script has been updated to allow several of its parameters
762 to be set from environment variables. This also enables a custom run
763 directory of Asterisk to be specified, instead of defaulting to /tmp.
765 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
766 its value to determine the directory to assume is the top-level directory of
767 the source tree. If the variable is not set, it defaults to the current
768 behavior and uses the current working directory.
770 ------------------------------------------------------------------------------
771 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
772 ------------------------------------------------------------------------------
776 * Asterisk now has protocol independent support for processing text messages
777 outside of a call. Messages are routed through the Asterisk dialplan.
778 SIP MESSAGE and XMPP are currently supported. There are options in
779 jabber.conf and sip.conf to allow enabling these features.
780 -> jabber.conf: see the "sendtodialplan" and "context" options.
781 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
782 and "outofcall_message_context" options.
783 The MESSAGE() dialplan function and MessageSend() application have been
784 added to go along with this functionality. More detailed usage information
785 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
786 * If real-time text support (T.140) is negotiated, it will be preferred for
787 sending text via the SendText application. For example, via SIP, messages
788 that were once sent via the SIP MESSAGE request would be sent via RTP if
789 T.140 text is negotiated for a call.
793 * parkedmusicclass can now be set for non-default parking lots.
795 Asterisk Manager Interface
796 --------------------------
797 * PeerStatus now includes Address and Port.
798 * Added Hold events for when the remote party puts the call on and off hold
799 for chan_dahdi ISDN channels.
800 * Added new action MeetmeListRooms to list active conferences (shows same
801 data as "meetme list" at the CLI).
802 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
803 Description field that is set by 'description' in the channel configuration
805 * Added Uniqueid header to UserEvent.
806 * Added new action FilterAdd to control event filters for the current session.
807 This requires the system permission and uses the same filter syntax as
808 filters that can be defined in manager.conf
809 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
810 versions had some instances of the event converted, but others were left
811 as-is. All Unlink events should now be converted to Bridge events. The AMI
812 protocol version number was incremented to 1.2 as a result of this change.
815 --------------------------
816 * The HTTP Server can bind to IPv6 addresses.
819 --------------------------
820 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
821 with busydetect. usage example: busypattern=200,200,200,600
824 --------------------------
825 * New 'gtalk show settings' command showing the current settings loaded from
827 * The 'logger reload' command now supports an optional argument, specifying an
828 alternate configuration file to use.
829 * 'dialplan add extension' command will now automatically create a context if
830 the specified context does not exist with a message indicated it did so.
831 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
832 Description field which can be populated with 'description' in the channel
833 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
836 --------------------------
837 * The filter option in cdr_adaptive_odbc now supports negating the argument,
838 thus allowing records which do NOT match the specified filter.
839 * Added ability to log CONGESTION calls to CDR
842 --------------------------
843 * Ability to define custom SILK formats in codecs.conf.
844 * Addition of speex32 audio format with translation.
845 * CELT codec pass-through support and ability to define
846 custom CELT formats in codecs.conf.
847 * Ability to read raw signed linear files with sample rates
848 ranging from 8khz - 192khz. The new file extensions introduced
849 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
850 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
851 Skinny, H.323, etc) can still only support the following codecs:
852 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
853 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
854 Video: h261, h263, h263p, h264, mpeg4
859 --------------------------
860 * New highly optimized and customizable ConfBridge application capable of
861 mixing audio at sample rates ranging from 8khz-96khz.
862 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
863 and bridge profiles on a channel.
864 * CONFBRIDGE_INFO dialplan function capable of retrieving information
865 about a conference such as locked status and number of parties, admins,
867 * Addition of video_mode option in confbridge.conf for adding video support
868 into a bridge profile.
869 * Addition of the follow_talker video_mode in confbridge.conf. This video
870 mode dynamically switches the video feed to always display the loudest talker
871 supplying video in the conference.
875 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
876 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
877 variables from asterisk.conf.
881 * Addition of the JITTERBUFFER dialplan function. This function allows
882 for jitterbuffering to occur on the read side of a channel. By using
883 this function conference applications such as ConfBridge and MeetMe can
884 have the rx streams jitterbuffered before conference mixing occurs.
885 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
887 * Added STRREPLACE function. This function let's the user search a variable
888 for a given string to replace with another string as many times as the
889 user specifies or just throughout the whole string.
890 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
891 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
892 * Added extensions to chan_ooh323 in function CHANNEL()
894 libpri channel driver (chan_dahdi) DAHDI changes
895 --------------------------
896 * Added moh_signaling option to specify what to do when the channel's bridged
897 peer puts the ISDN channel on hold.
898 * Added display_send and display_receive options to control how the display ie
899 is handled. To send display text from the dialplan use the SendText()
900 application when the option is enabled.
901 * Added mcid_send option to allow sending a MCID request on a span.
904 --------------------------
905 * Added setvar option to calendar.conf to allow setting channel variables on
906 notification channels.
907 * Added "calendar show types" CLI command to list registered calendar
911 --------------------------
912 * Added two new options, r and t with file name arguments to record
913 single direction (unmixed) audio recording separate from the bidirectional
914 (mixed) recording. The mixed file name argument is optional now as long
915 as at least one recording option is used.
918 --------------------------
919 * Added a new option, l, which will disable local call optimization for
920 channels involved with the FollowMe thread. Use this option to improve
921 compatability for a FollowMe call with certain dialplan apps, options, and
925 --------------------------
926 * Added option "k" that will automatically close the conference when there's
927 only one person left when a user exits the conference.
930 --------------------------
931 * cel_pgsql now supports the 'extra' column for data added using the
932 CELGenUserEvent() application.
935 --------------------------
936 * Support for defining hints has been added to pbx_lua. See the 'hints' table
937 in the sample extensions.lua file for syntax details.
938 * Applications that perform jumps in the dialplan such as Goto will now
939 execute properly. When pbx_lua detects that the context, extension, or
940 priority we are executing on has changed it will immediately return control
941 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
942 the priority after the currently executing priority.
943 * An autoservice is now started by default for pbx_lua channels. It can be
944 stopped and restarted using the autoservice_stop() and autoservice_start()
948 --------------------------
949 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
950 into a FAXStatus event with an 'Operation' header that will be either
951 'send', 'receive', and 'gateway'.
952 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
953 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
954 feature will handle converting a fax call between an audio T.30 fax terminal
955 and an IFP T.38 fax terminal.
959 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
960 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
961 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
965 * Added general option negative_penalty_invalid default off. when set
966 members are seen as invalid/logged out when there penalty is negative.
967 for realtime members when set remove from queue will set penalty to -1.
968 * Added queue option autopausedelay when autopause is enabled it will be
969 delayed for this number of seconds since last successful call if there
970 was no prior call the agent will be autopaused immediately.
971 * Added member option ignorebusy this when set and ringinuse is not
972 will allow per member control of multiple calls as ringinuse does for
977 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
979 * Added 'k' option to MeetMe to automatically kill the conference when there's only
980 one participant left (much like a normal call bridge)
981 * Added extra argument to Originate to set timeout.
985 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
986 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
987 utility in the UTILS section of menuselect. If an existing astdb is found and no
988 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
989 convert an existing astdb to the SQLite3 version automatically at runtime.
993 * Modules marked as deprecated are no longer marked as building by default. Enabling
994 these modules is still available via menuselect.
998 * authdebug is now disabled by default. To enable this functionaility again
999 set authdebug = yes in iax.conf.
1003 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
1004 releases it was disabled.
1008 * The PBX core previously made a call with a non-existing extension test for
1009 extension s@default and jump there if the extension existed.
1010 This was a bad default behaviour and violated the principle of least surprise.
1011 It has therefore been changed in this release. It may affect some
1012 applications and configurations that rely on this behaviour. Most channel
1013 drivers have avoided this for many releases by testing whether the extension
1014 called exists before starting the PBX and generating a local error.
1015 This behaviour still exists and works as before.
1017 Extension "s" is used when no extension is given in a channel driver,
1018 like immediate answer in DAHDI or calling to a domain with no user part
1021 ------------------------------------------------------------------------------
1022 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
1023 ------------------------------------------------------------------------------
1027 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
1028 now defaults to force_rport. It is very important that phones requiring nat=no be
1029 specifically set as such instead of relying on the default setting. If at all
1030 possible, all devices should have nat settings configured in the general section as
1031 opposed to configuring nat per-device.
1032 * Added preferred_codec_only option in sip.conf. This feature limits the joint
1033 codecs sent in response to an INVITE to the single most preferred codec.
1034 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
1035 to be used for the outgoing call. It must be one of the codecs configured
1037 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
1038 to be used for holding a private key. If tlsprivatekey is not specified,
1039 tlscertfile is searched for both public and private key.
1040 * Added tlsclientmethod option to sip.conf. This allows the protocol for
1041 outbound client connections to be specified.
1042 * The sendrpid parameter has been expanded to include the options
1043 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
1044 header to be sent (equivalent to setting sendrpid=yes) and setting
1045 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
1046 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
1047 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
1048 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
1049 will accept the SDP even if the SDP version number is not properly incremented,
1050 but will generate a warning in the log indicating that the SIP peer that sent
1051 the SDP should have the 'ignoresdpversion' option set.
1052 * The 'nat' option has now been been changed to have yes, no, force_rport, and
1053 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
1054 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
1055 remote side requests it and disables symmetric RTP support. Setting it to
1056 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
1057 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
1058 and enables symmetric RTP support.
1059 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
1060 response. This permits the master channel to know how each channel dialled
1061 in a multi-channel setup resolved in an individual way. This carries a
1062 performance penalty and can be disabled in sip.conf using the
1063 'storesipcause' option.
1064 * Added 'externtcpport' and 'externtlsport' options to allow custom port
1065 configuration for the externip and externhost options when tcp or tls is used.
1066 * Added support for message body (stored in content variable) to SIP NOTIFY message
1067 accessible via AMI and CLI.
1068 * Added 'media_address' configuration option which can be used to explicitly specify
1069 the IP address to use in the SDP for media (audio, video, and text) streams.
1070 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
1071 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
1073 * Added 'use_q850_reason' configuration option for generating and parsing
1074 if available Reason: Q.850;cause=<cause code> header. It is implemented
1075 in some gateways for better passing PRI/SS7 cause codes via SIP.
1076 * When dialing SIP peers, a new component may be added to the end of the dialstring
1077 to indicate that a specific remote IP address or host should be used when dialing
1078 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
1079 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
1080 ability to selectively force bridged channels to also be encrypted is also
1081 implemented. Branching in the dialplan can be done based on whether or not
1082 a channel has secure media and/or signaling.
1083 * Added directmediapermit/directmediadeny to limit which peers can send direct media
1085 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
1086 Charge messages to snom phones.
1087 * Added support for G.719 media streams.
1088 * Added support for 16khz signed linear media streams.
1089 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
1090 RTP has been outfitted with the same abilities.
1091 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
1092 available in device configurations as well as in the dial plan.
1093 * Addition of the 'subscribe_network_change' option for turning on and off
1094 res_stun_monitor module support in chan_sip.
1095 * Addition of the 'auth_options_requests' option for turning on and off
1096 authentication for OPTIONS requests in chan_sip.
1100 * Add #tryinclude statement for config files. This provides the same
1101 functionality as the #include statement however an asterisk module will
1102 still load if the filename does not exist. Using the #include statement
1103 Asterisk will not allow the module to load.
1107 * Added rtsavesysname option into iax.conf to allow the systname to be saved
1108 on realtime updates.
1109 * Added the ability for chan_iax2 to inform the dialplan whether or not
1110 encryption is being used. This interoperates with the SIP SRTP implementation
1111 so that a secure SIP call can be bridged to a secure IAX call when the
1112 dialplan requires bridged channels to be "secure".
1113 * Addition of the 'subscribe_network_change' option for turning on and off
1114 res_stun_monitor module support in chan_iax.
1119 * Added ability to preset channel variables on indicated lines with the setvar
1120 configuration option. Also, clearvars=all resets the list of variables back
1122 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
1123 See configs/res_pktccops.conf for more information.
1125 XMPP Google Talk/Jingle changes
1126 -------------------------------
1127 * Added the externip option to gtalk.conf.
1128 * Added the stunaddr option to gtalk.conf which allows for the automatic
1129 retrieval of the external ip from a stun server.
1133 * Added 'p' option to PickupChan() to allow for picking up channel by the first
1134 match to a partial channel name.
1135 * Added .m3u support for Mp3Player application.
1136 * Added progress option to the app_dial D() option. When progress DTMF is
1137 present, those values are sent immediately upon receiving a PROGRESS message
1138 regardless if the call has been answered or not.
1139 * Added functionality to the app_dial F() option to continue with execution
1140 at the current location when no parameters are provided.
1141 * Added the 'a' option to app_dial to answer the calling channel before any
1142 announcements or macros are executed.
1143 * Modified app_dial to set answertime when the called channel answers even if
1144 the called channel hangs up during playback of an announcement.
1145 * Modified app_dial 'r' option to support an additional parameter to play an
1146 indication tone from indications.conf
1147 * Added c() option to app_chanspy. This option allows custom DTMF to be set
1148 to cycle through the next available channel. By default this is still '*'.
1149 * Added x() option to app_chanspy. This option allows DTMF to be set to
1150 exit the application.
1151 * The Voicemail application has been improved to automatically ignore messages
1152 that only contain silence.
1153 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
1154 associated mailbox(es) to be greetings-only.
1155 * The ChanSpy application now has the 'S' option, which makes the application
1156 automatically exit once it hits a point where no more channels are available
1158 * The ChanSpy application also now has the 'E' option, which spies on a single
1159 channel and exits when that channel hangs up.
1160 * The MeetMe application now turns on the DENOISE() function by default, for
1161 each participant. In our tests, this has significantly decreased background
1162 noise (especially noisy data centers).
1163 * Voicemail now permits storage of secrets in a separate file, located in the
1164 spool directory of each individual user. The control for this is located in
1165 the "passwordlocation" option in voicemail.conf. Please see the sample
1166 configuration for more information.
1167 * The ChanIsAvail application now exposes the returned cause code using a separate
1168 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
1169 * Added 'd' option to app_followme. This option disables the "Please hold"
1171 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
1172 received will terminate recording.
1173 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
1174 Previously the folder could only be set per context, but has now been extended
1175 using the imapfolder option.
1176 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
1177 * Voicemail now allows the pager date format to be specified separately from the
1179 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
1180 to allow joining, leaving, and sending text to group chats.
1181 * MeetMe has a new option 'G' to play an announcement before joining a conference.
1182 * Page has a new option 'A(x)' which will playback an announcement simultaneously
1183 to all paged phones (and optionally excluding the caller's one using the new
1184 option 'n') before the call is bridged.
1185 * The 'f' option to Dial has been augmented to take an optional argument. If no
1186 argument is provided, the 'f' option works as it always has. If an argument is
1187 provided, then the connected party information of all outgoing channels created
1188 during the Dial will be set to the argument passed to the 'f' option.
1189 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
1191 * The OSP lookup application adds in/outbound network ID, optional security,
1192 number portability, QoS reporting, destination IP port, custom info and service
1194 * Added new application VMSayName that will play the recorded name of the voicemail
1195 user if it exists, otherwise will play the mailbox number.
1196 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
1197 retrieve state for a particular bridge, where <name> is the conference name
1198 * app_directory now allows exiting at any time using the operator or pound key.
1199 * Voicemail now supports setting a locale per-mailbox.
1200 * Two new applications are provided for declining counting phrases in multiple
1201 languages. See the application notes for SayCountedNoun and SayCountedAdj for
1203 * Voicemail now runs the externnotify script when pollmailboxes is activated and
1205 * Voicemail now includes rdnis within msgXXXX.txt file.
1206 * ExternalIVR now supports IPv6 addresses.
1207 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
1208 at https://wiki.asterisk.org/wiki/x/oQBB
1209 * ParkedCall and Park can now specify the parking lot to use.
1213 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
1214 over SRV records associated with a specific service. From the CLI, type
1215 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
1216 details on how these may be used.
1217 * PITCH_SHIFT dialplan function added. This function can be used to modify the
1218 pitch of a channel's tx and rx audio streams.
1219 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
1220 setting various connected line and redirecting party information.
1221 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
1222 support ISDN subaddressing.
1223 * The CHANNEL() function now supports the "name" and "checkhangup" options.
1224 * For DAHDI channels, the CHANNEL() dialplan function now allows
1225 the dialplan to request changes in the configuration of the active
1226 echo canceller on the channel (if any), for the current call only.
1229 exten => s,n,Set(CHANNEL(echocan_mode)=off)
1231 The possible values are:
1233 on - normal mode (the echo canceller is actually reinitialized)
1235 fax - FAX/data mode (NLP disabled if possible, otherwise completely
1237 voice - voice mode (returns from FAX mode, reverting the changes that
1238 were made when FAX mode was requested)
1239 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
1240 and setting variables on the channel which created the current channel.
1241 Administrators should take care to avoid naming conflicts, when multiple
1242 channels are dialled at once, especially when used with the Local channel
1243 construct (which all could set variables on the master channel). Usage
1244 of the HASH() dialplan function, with the key set to the name of the slave
1245 channel, is one approach that will avoid conflicts.
1246 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
1248 * func_odbc now allows multiple row results to be retrieved without using
1249 mode=multirow. If rowlimit is set, then additional rows may be retrieved
1250 from the same query by using the name of the function which retrieved the
1251 first row as an argument to ODBC_FETCH().
1252 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
1253 dialplan. This function returns the content of the received message.
1254 * Added REPLACE, which searches a given variable name for a set of characters,
1255 then either replaces them with a single character or deletes them.
1256 * Added PASSTHRU, which literally passes the same argument back as its return
1257 value. The intent is to be able to use a literal string argument to
1258 functions that currently require a variable name as an argument.
1259 * HASH-associated variables now can be inherited across channel creation, by
1260 prefixing the name of the hash at assignment with the appropriate number of
1261 underscores, just like variables.
1262 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
1263 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
1264 whether or not channels that are bridged to the current channel will be
1265 required to have secure signaling and/or media.
1266 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
1267 the current channel has secure signaling and/or media.
1268 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
1269 "no_media_path" option.
1270 Returns "0" if there is a B channel associated with the call.
1271 Returns "1" if no B channel is associated with the call. The call is either
1272 on hold or is a call waiting call.
1273 * Added option to dialplan function CDR(), the 'f' option
1274 allows for high resolution times for billsec and duration fields.
1275 * FILE() now supports line-mode and writing.
1276 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
1277 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
1281 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
1282 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
1283 and is set when a dynamic feature is triggered.
1284 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
1285 to dynamically create a new parking lot matching the value this varible is
1287 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
1288 features.conf that should be the base for dynamic parkinglots.
1289 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
1290 parkinglot should have.
1291 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
1292 parkinglot should have.
1293 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
1298 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
1299 timeout has expired.
1300 * Added 'R' option to app_queue. This option stops moh and indicates ringing
1301 to the caller when an Agent's phone is ringing. This can be used to indicate
1302 to the caller that their call is about to be picked up, which is nice when
1303 one has been on hold for an extened period of time.
1304 * A new config option, penaltymemberslimit, has been added to queues.conf.
1305 When set this option will disregard penalty settings when a queue has too
1307 * A new option, 'I' has been added to both app_queue and app_dial.
1308 By setting this option, Asterisk will not update the caller with
1309 connected line changes or redirecting party changes when they occur.
1310 * A 'relative-periodic-announce' option has been added to queues.conf. When
1311 enabled, this option will cause periodic announce times to be calculated
1312 from the end of announcements rather than from the beginning.
1313 * The autopause option in queues.conf can be passed a new value, "all." The
1314 result is that if a member becomes auto-paused, he will be paused in all
1315 queues for which he is a member, not just the queue that failed to reach
1317 * Added dialplan function QUEUE_EXISTS to check if a queue exists
1318 * The queue logger now allows events to optionally propagate to a file,
1319 even when realtime logging is turned on. Additionally, realtime logging
1320 supports sending the event arguments to 5 individual fields, although it
1321 will fallback to the previous data definition, if the new table layout is
1324 mISDN channel driver (chan_misdn) changes
1325 ----------------------------------------
1326 * Added display_connected parameter to misdn.conf to put a display string
1327 in the CONNECT message containing the connected name and/or number if
1328 the presentation setting permits it.
1329 * Added display_setup parameter to misdn.conf to put a display string
1330 in the SETUP message containing the caller name and/or number if the
1331 presentation setting permits it.
1332 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
1333 indicate the dialplan settings are to be obtained from the asterisk
1335 * Made misdn.conf parameter callerid accept the "name" <number> format
1336 used by the rest of the system.
1337 * Made use the nationalprefix and internationalprefix misdn.conf
1338 parameters to prefix any received number from the ISDN link if that
1339 number has the corresponding Type-Of-Number. NOTE: This includes
1340 comparing the incoming call's dialed number against the MSN list.
1341 * Added the following new parameters: unknownprefix, netspecificprefix,
1342 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
1343 received number from the ISDN link if that number has the corresponding
1345 * Added new dialplan application misdn_command which permits controlling
1346 the CCBS/CCNR functionality.
1347 * Added new dialplan function mISDN_CC which permits retrieval of various
1348 values from an active call completion record.
1349 * For PTP, you should manually send the COLR of the redirected-to party
1350 for an incomming redirected call if the incoming call could experience
1351 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
1352 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
1353 if the REDIRECTING(from-num) is not empty.
1354 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
1355 option on all of the REDIRECTING statements before dialing the
1356 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
1357 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
1358 redirecting-to presentation (COLR) when it becomes available.
1359 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
1362 thirdparty mISDN enhancements
1363 -----------------------------
1364 mISDN has been modified by Digium, Inc. to greatly expand facility message
1366 * Enhanced COLP support for call diversion and transfer.
1367 * CCBS/CCNR support.
1369 The latest modified mISDN v1.1.x based version is available at:
1370 http://svn.digium.com/svn/thirdparty/mISDN/trunk
1371 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
1373 Tagged versions of the modified mISDN code are available under:
1374 http://svn.digium.com/svn/thirdparty/mISDN/tags
1375 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
1377 libpri channel driver (chan_dahdi) DAHDI changes
1378 -------------------------------------------
1379 * The channel variable PRIREDIRECTREASON is now just a status variable
1380 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
1381 to read and alter the reason.
1382 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
1383 redirected-to party for an incomming redirected call if the incoming call
1384 could experience further redirects. Just set the
1385 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
1386 to the COLR. A call has been redirected if the REDIRECTING(count) is not
1388 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
1389 use the inhibit(i) option on all of the REDIRECTING statements before
1390 dialing the redirected-to party. You still have to set the
1391 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
1392 will update the redirecting-to presentation (COLR) when it becomes available.
1393 * Added the ability to ignore calls that are not in a Multiple Subscriber
1394 Number (MSN) list for PTMP CPE interfaces.
1395 * Added dynamic range compression support for dahdi channels. It is
1396 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
1397 * Added support for ISDN calling and called subaddress with partial support
1398 for connected line subaddress.
1399 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
1400 * Added handling of received HOLD/RETRIEVE messages and the optional ability
1401 to transfer a held call on disconnect similar to an analog phone.
1402 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
1403 Will reroute/deflect an outgoing call when receive the message.
1404 Can use the DAHDISendCallreroutingFacility to send the message for the
1406 * Added standard location to add options to chan_dahdi dialing:
1407 Dial(DAHDI/g1[/extension[/options]])
1410 R Reverse charging indication
1411 * Added Reverse Charging Indication (Collect calls) send/receive option.
1412 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
1413 Dial(DAHDI/g1/extension/R)
1414 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
1415 (requires latest LibPRI)
1416 * Added ability to send/receive keypad digits in the SETUP message.
1417 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
1418 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
1419 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
1420 (requires latest LibPRI)
1421 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
1422 to eliminate tromboned calls. A tromboned call goes out an interface and comes
1423 back into the same interface. Tromboned calls happen because of call routing,
1424 call deflection, call forwarding, and call transfer.
1425 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
1426 * Added the ability to support call waiting calls. (The SETUP has no B channel
1428 * Added Malicious Call ID (MCID) event to the AMI call event class.
1429 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
1431 Asterisk Manager Interface
1432 --------------------------
1433 * The Hangup action now accepts a Cause header which may be used to
1434 set the channel's hangup cause.
1435 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
1436 to specify a separate .pem file to hold a private key. By default sslcert
1437 is used to hold both the public and private key.
1438 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
1439 for options containing the 'tls' prefix. For example, 'sslenable' is now
1440 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
1441 across all .conf files. All affected sample.conf files have been modified to
1442 reflect this change. Previous options such as 'sslenable' still work,
1443 but options with the 'tls' prefix are preferred.
1444 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
1445 in a channel. (res_mutestream.so)
1446 * The configuration file manager.conf now supports a channelvars option, which
1447 specifies a list of channel variables to include in each channel-oriented
1449 * The redirect command now has new parameters ExtraContext, ExtraExtension,
1450 and ExtraPriority to allow redirecting the second channel to a different
1451 location than the first.
1452 * Added new event "JabberStatus" in the Jabber module to monitor buddies
1454 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
1455 in a MixMonitor recording.
1456 * The 'iax2 show peers' output is now similar to the expected output of
1458 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
1460 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
1461 AOC-E messages on a channel.
1462 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
1463 conform more closely to similar events.
1464 * Added a new eventfilter option per user to allow whitelisting and blacklisting
1466 * Added optional parkinglot variable for park command.
1467 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
1468 if CallerIDNum and CallerIDName headers are also present.
1470 Channel Event Logging
1471 ---------------------
1472 * A new interface, CEL, is introduced here. CEL logs single events, much like
1473 the AMI, but it differs from the AMI in that it logs to db backends much
1474 like CDR does; is based on the event subsystem introduced by Russell, and
1475 can share in all its benefits; allows multiple backends to operate like CDR;
1476 is specialized to event data that would be of concern to billing sytems,
1477 like CDR. Backends for logging and accounting calls have been produced,
1478 but a new CDR backend is still in development.
1482 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
1483 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
1484 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
1485 * Multiple files and formats can now be specified in cdr_custom.conf.
1486 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
1487 See configs/cdr_syslog.conf.sample for more information.
1488 * A 'sequence' field has been added to CDRs which can be combined with
1489 linkedid or uniqueid to uniquely identify a CDR.
1490 * Handling of billsec and duration field has changed. If your table definition
1491 specifies those fields as float,double or similar they will now be logged with
1492 microsecond accuracy instead of a whole integer.
1494 Calendaring for Asterisk
1495 ------------------------
1496 * A new set of modules were added supporing calendar integration with Asterisk.
1497 Dialplan functions for reading from and writing to calendars are included,
1498 as well as the ability to execute dialplan logic upon calendar event notifications.
1499 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
1500 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
1501 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
1502 2003 support does not support forms-based authentication).
1504 Call Completion Supplementary Services for Asterisk
1505 ---------------------------------------------------
1506 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
1507 DAHDI/ISDN supports call completion for the following switch types:
1508 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
1509 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
1511 Multicast RTP Support
1512 ---------------------
1513 * A new RTP engine and channel driver have been added which supports Multicast RTP.
1514 The channel driver can be used with the Page application to perform multicast RTP
1515 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
1516 Type can be either basic or linksys.
1517 Destination is the IP address and port for the RTP packets.
1518 Control address is specific to the linksys type and is used for sending the control
1519 packets unique to them.
1521 Security Events Framework
1522 -------------------------
1523 * Asterisk has a new C API for reporting security events. The module res_security_log
1524 sends these events to the "security" logger level. Currently, AMI is the only
1525 Asterisk component that reports security events. However, SIP support will be
1526 coming soon. For more information on the security events framework, see the
1527 "Asterisk Security Framework" section of the Asterisk wiki at
1528 https://wiki.asterisk.org/wiki/x/wgBQ
1529 * SIP support was added in Asterisk 10
1530 * This API now supports IPv6 addresses
1534 * A technology independent fax frontend (res_fax) has been added to Asterisk.
1535 * A spandsp based fax backend (res_fax_spandsp) has been added.
1536 * The app_fax module has been deprecated in favor of the res_fax module and
1537 the new res_fax_spandsp backend.
1538 * The SendFAX and ReceiveFAX applications now send their log messages to a
1539 'fax' logger level, instead of to the generic logger levels. To see these
1540 messages, the system's logger.conf file will need to direct the 'fax' logger
1541 level to one or more destinations; the logger.conf.sample file includes an
1542 example of how to do this. Note that if the 'fax' logger level is *not*
1543 directed to at least one destination, log messages generated by these
1544 applications will be lost, and that if the 'fax' logger level is directed to
1545 the console, the 'core set verbose' and 'core set debug' CLI commands will
1546 have no effect on whether the messages appear on the console or not.
1550 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
1551 Now, in order to enable transmitting silence during record the transmit_silence
1552 option should be used. transmit_silence_during_record remains a valid option, but
1553 defaults to the behavior of the transmit_silence option.
1554 * Addition of the Unit Test Framework API for managing registration and execution
1555 of unit tests with the purpose of verifying the operation of C functions.
1556 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
1557 XMPP text messages to the remote JID.
1558 * Modules.conf has a new option - "require" - that marks a module as critical for
1559 the execution of Asterisk.
1560 If one of the required modules fail to load, Asterisk will exit with a return
1562 * An 'X' option has been added to the asterisk application which enables #exec support.
1563 This allows #exec to be used in asterisk.conf.
1564 * jabber.conf supports a new option auth_policy that toggles auto user registration.
1565 * A new lockconfdir option has been added to asterisk.conf to protect the
1566 configuration directory (/etc/asterisk by default) during reloads.
1567 * The parkeddynamic option has been added to features.conf to enable the creation
1568 of dynamic parkinglots.
1569 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
1570 the reportalarms config option.
1571 * chan_dahdi supports dialing configuring and dialing by device file name.
1572 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
1573 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
1574 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
1575 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
1576 Handy for the above name-based syntax as it does not depend on
1577 initialization order.
1578 * The Realtime dialplan switch now caches entries for 1 second. This provides a
1579 significant increase in performance (about 3X) for installations using this switchtype.
1580 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
1581 AIS. For more information, please see the Distributed Device State section of the
1582 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1583 * The addition of G.719 pass-through support.
1584 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
1585 during device configuration.
1586 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
1587 have less than 3 lines on the LCD.
1588 * Realtime now supports database failover. See the sample extconfig.conf for details.
1589 * The addition of improved translation path building for wideband codecs. Sample
1590 rate changes during translation are now avoided unless absolutely necessary.
1591 * The addition of the res_stun_monitor module for monitoring and reacting to network
1592 changes while behind a NAT.
1593 * DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf.
1594 DTMF Valid/Invalid number of hits/misses can be set in dsp.conf.
1595 These allow support for any Administration. Default is AT&T values.
1599 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
1600 optionally accept a filename, to apply the setting only to the code generated from
1601 that source file when Asterisk was built. However, there are some modules in Asterisk
1602 that are composed of multiple source files, so this did not result in the behavior
1603 that users expected. In this version, 'core set debug' and 'core set verbose'
1604 can optionally accept *module* names instead (with or without the .so extension),
1605 which applies the setting to the entire module specified, regardless of which source
1606 files it was built from.
1607 * New 'manager show settings' command showing the current settings loaded from
1609 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
1610 the channel hangup request to all channels.
1611 * Added a "core reload" CLI command that executes a global reload of Asterisk.
1613 ------------------------------------------------------------------------------
1614 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
1615 ------------------------------------------------------------------------------
1619 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
1620 Snom phones use this for call pickup of extensions that the phone is
1622 * Added support for setting the domain in the URI for caller of an
1623 outbound call by using the SIPFROMDOMAIN channel variable.
1624 * Added a new configuration option "remotesecret" for authentication to
1625 remote services. For backwards compatibility, "secret" still has the
1626 same function as before, but now you can configure both a remote secret and a
1627 local secret for mutual authentication.
1628 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
1629 the sound will be played to the target of an attended transfer
1630 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
1631 finer control over how many peers Asterisk will qualify and the gap between them
1632 when all peers need to be qualified at the same time.
1633 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
1634 (either globally or for a specific peer), chan_sip will treat any SDP data
1635 it receives as new data and update the media stream accordingly. By
1636 default, Asterisk will only modify the media stream if the SDP session
1637 version received is different from the current SDP session version. This
1638 option is required to interoperate with devices that have non-standard SDP
1639 session version implementations (observed with Microsoft OCS). This option
1640 is disabled by default.
1641 * The parsing of register => lines in sip.conf has been modified to allow a port
1642 to be present in the "user" portion. Please see the sip.conf.sample file for more
1644 * Added support for subscribing to MWI on a remote server and making the status available
1645 as a mailbox. Please see the sip.conf.sample file for more information.
1646 * Added a function to remove SIP headers added in the dialplan before the
1647 first INVITE is generated - SIPRemoveHeader()
1648 * Channel variables set with setvar= in a device configuration is now
1649 set both for inbound and outbound calls.
1650 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
1654 * Added immediate option to iax.conf
1655 * Added forceencryption option to iax.conf
1656 * Added Encryption and Trunk status to manager command "iaxpeers"
1660 * The configuration file now holds separate sections for devices and lines.
1661 Please have a look at configs/skinny.conf.sample and change your skinny.conf
1666 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
1667 support for LibOpenR2. http://www.libopenr2.org/
1668 * The UK option waitfordialtone has been added for use with BT analog
1670 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
1671 is used in conjunction with the 'faxdetect' configuration option. When
1672 'faxbuffers' is used and fax tones are detected, the channel will dynamically
1673 switch to the configured faxbuffers policy. For example, to use 6 buffers
1674 and a 'full' buffer policy for a fax transmission, add:
1676 The faxbuffers configuration will be in affect until the call is torn down.
1677 * Added service message support for 4ESS/5ESS switches.
1681 * For DAHDI channels, the CHANNEL() dialplan function now
1682 supports changing the channel's buffer policy (for the current
1683 call only), using this syntax:
1685 exten => s,n,Set(CHANNEL(buffers)=6,full)
1687 This would change the channel to the 'full' buffer policy and
1688 6 (six) buffers. Possible options for this setting are the same
1689 as those in chan_dahdi.conf.
1690 * Added a new dialplan function, CURLOPT, which permits setting various
1691 options that may be useful with the CURL dialplan function, such as
1692 cookies, proxies, connection timeouts, passwords, etc.
1693 * Permit the syntax and synopsis fields of the corresponding dialplan
1694 functions to be individually set from func_odbc.conf.
1695 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
1696 * func_odbc now may specify an insert query to execute, when the write query
1697 affects 0 rows (usually indicating that no such row exists).
1698 * Added a new dialplan function, LISTFILTER, which permits removing elements
1699 from a set list, by name. Uses the same general syntax as the existing CUT
1700 and FIELDQTY dialplan functions, which also manage lists.
1701 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
1702 obtaining realtime data from the dialplan.
1703 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
1704 a subroutine when using the GoSub() and Return() applications.
1705 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
1706 of "core show function AUDIOHOOK_INHERIT" from the CLI
1707 * Added AES_ENCRYPT. For information on its use, please see the output
1708 of "core show function AES_ENCRYPT" from the CLI
1709 * Added AES_DECRYPT. For information on its use, please see the output
1710 of "core show function AES_DECRYPT" from the CLI
1711 * func_odbc now supports database transactions across multiple queries.
1715 * Scheduled meetme conferences may now have their end times extended by
1717 * app_authenticate now gives the ability to select a prompt other than
1719 * app_directory now pays attention to the searchcontexts setting in
1720 voicemail.conf and will look through all contexts, if no context is
1721 specified in the initial argument.
1722 * A new application, Originate, has been introduced, that allows asynchronous
1723 call origination from the dialplan.
1724 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
1725 in addition to the setting in the "general" context.
1726 * Added ConfBridge dialplan application which does conference bridges without
1727 DAHDI. For information on its use, please see the output of
1728 "core show application ConfBridge" from the CLI.
1732 * The Asterisk CLI has a new command, "channel redirect", which is similar in
1733 operation to the AMI Redirect action.
1734 * extensions.conf now allows you to use keyword "same" to define an extension
1735 without actually specifying an extension. It uses exactly the same pattern
1736 as previously used on the last "exten" line. For example:
1737 exten => 123,1,NoOp(something)
1738 same => n,SomethingElse()
1739 * musiconhold.conf classes of type 'files' can now use relative directory paths,
1740 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
1741 * All deprecated CLI commands are removed from the sourcecode. They are now handled
1742 by the new clialiases module. See cli_aliases.conf.sample file.
1743 * Times within timespecs are now accurate down to the minute. This is a change
1744 from historical Asterisk, which only provided timespecs rounded to the nearest
1745 even (read: evenly divisible by 2) minute mark.
1746 * The realtime switch now supports an option flag, 'p', which disables searches for
1748 * In addition to a time range and date range, timespecs now accept a 5th optional
1749 argument, timezone. This allows you to perform time checks on alternate
1750 timezones, especially if those daylight savings time ranges vary from your
1751 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
1753 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
1754 give you the correct output for an asterisk box behind nat. It will give you the
1755 externhost and localnet settings.
1756 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
1757 can connect calls in passthrough mode, as well as record and play back files.
1758 * Successful and unsuccessful call pickup can now be alerted through sounds, by
1759 using pickupsound and pickupfailsound in features.conf.
1760 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
1761 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
1762 instead of the /var/run/asterisk.pid where it used to be. This will make
1763 installs as non-root easier to manage.
1768 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
1769 be written; they will no longer be explicitly written.
1771 Asterisk Manager Interface
1772 --------------------------
1773 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
1774 a non-empty value) in your request. If you do this, any pending AMI events will
1775 *not* be included in the response to your request as they would normally, but
1776 will be left in the event queue for the next request you make to retrieve. For
1777 some applications, this will allow you to guarantee that you will only see
1778 events in responses to 'WaitEvent' actions, and can better know when to expect them.
1779 To know whether the Asterisk server supports this header or not, your client can
1780 inspect the first response back from the server to see if it includes this header:
1782 Pragma: SuppressEvents
1784 If this is included, the server supports event suppression.
1786 * Added 4 new Actions to list skinny device(s) and line(s)
1792 LDAP Schema File Additions
1793 --------------------------
1794 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
1795 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
1797 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
1798 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
1799 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
1800 * Removed redundant IPaddr (there's already IPAddress)
1801 - Gives more configuration Flags for SIP-Users available (tested)
1802 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
1803 without extensibleObject (which really should be the last resort); gives
1804 also additional possibilities for LDAP-filter
1806 ------------------------------------------------------------------------------
1807 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
1808 ------------------------------------------------------------------------------
1810 Device State Handling
1811 ---------------------
1812 * The event infrastructure in Asterisk got another big update to help support
1813 distributed events. It currently supports distributed device state and
1814 distributed Voicemail MWI (Message Waiting Indication). A new module has
1815 been merged, res_ais, which facilitates communicating events between servers.
1816 It uses the SAForum AIS (Service Availability Forum Application Interface
1817 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
1818 a cluster of Asterisk servers, and to share events between them. For more
1819 information on setting this up, refer to the Distributed Device State section
1820 of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1824 * Added a new dialplan function, AST_CONFIG(), which allows you to access
1825 variables from an Asterisk configuration file.
1826 * The JACK_HOOK function now has a c() option to supply a custom client name.
1827 * Added two new dialplan functions from libspeex for audio gain control and
1828 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
1829 rx directions of a channel from the dialplan.
1830 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
1831 based on other parameters. The default is still to search based on the
1832 forwarding station ID. However, there are new options that allow you to search
1833 based on the message desk terminal ID, or the message desk number.
1834 * TIMEOUT() has been modified to be accurate down to the millisecond.
1835 * ENUM*() functions now include the following new options:
1836 - 'u' returns the full URI and does not strip off the URI-scheme.
1837 - 's' triggers ISN specific rewriting
1838 - 'i' looks for branches into an Infrastructure ENUM tree
1839 - 'd' for a direct DNS lookup without any flipping of digits.
1840 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
1841 * CHANNEL() now has options for the maximum, minimum, and standard or normal
1842 deviation of jitter, rtt, and loss for a call using chan_sip.
1844 DAHDI channel driver (chan_dahdi) Changes
1845 ----------------------------------------
1846 * Channels can now be configured using named sections in chan_dahdi.conf, just
1847 like other channel drivers, including the use of templates.
1848 * The default for pridialplan has changed from 'national' to 'unknown'.
1852 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
1853 to something that matches the pattern a hint will be created using the contents
1854 and variables evaluated.
1855 * Dialplan matching has been extended to allow an extension to return to the
1856 PBX core to wait for more digits. This is done by using the new dialplan
1857 application called "Incomplete". This will permit a whole new level of
1858 extension control, by giving the administrator more control over early
1859 matches employing one of the short-circuit pattern match operators. Note
1860 that custom applications can trigger this same behavior by returning the
1861 special value AST_PBX_INCOMPLETE.
1865 * Directory now permits both first and last names to be matched at the same
1866 time. In addition, the number of digits to enter of the name can be set in
1867 the arguments to Directory; previously, you could enter only 3, regardless
1868 of how many names are in your company. For large companies, this should be
1870 * Voicemail now permits a mailbox setting to wrap around from first to last
1871 messages, if the "messagewrap" option is set to a true value.
1872 * Voicemail now permits an external script to be run, for password validation.
1873 The script should output "VALID" or "INVALID" on stdout, depending upon the
1874 wish to validate or invalidate the password given. Arguments are:
1875 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
1877 * Dial has a new option: F(context^extension^pri), which permits a callee to
1878 continue in the dialplan, at the specified label, if the caller hangs up.
1879 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
1880 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
1881 * The Jack application now has a c() option to supply a custom client name.
1882 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
1883 like the pre-existing whisper mode, except that the spy can also talk to the
1884 participant on the bridged channel as well.
1885 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
1886 to be spoken instead of the channel name or number. For more information on the
1887 use of this option, issue the command "core show application ChanSpy" from the
1889 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
1890 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
1891 words, if using the 'd' option, it is not possible to enter a number to append to
1892 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
1893 change to whisper mode, and pressing 6 will change to barge mode.
1894 * ExternalIVR now takes several options that affect the way it performs, as
1895 well as having several new commands. Please see the External IVR page on the Asterisk
1896 wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
1897 * Added ability to communicate over a TCP socket instead of forking a child process for the
1898 ExternalIVR application.
1899 * ChanIsAvail has a new option, 'a', which will return all available channels instead
1900 of just the first one if you give the function more then one channel to check.
1901 * PrivacyManager now takes an option where you can specify a context where the
1902 given number will be matched. This way you have more control over who is allowed
1903 and it stops the people who blindly enter 10 digits.
1904 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
1905 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
1906 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
1907 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
1908 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
1909 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
1910 * The Dial() application no longer copies the language used by the caller to the callee's
1911 channel. If you desire for the caller's channel's language to be used for file playback
1912 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
1913 * SendImage() no longer hangs up the channel on error; instead, it sets the
1914 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
1915 'UNSUPPORTED'. This change makes SendImage() more consistent with other
1917 * Park has a new option, 's', which silences the announcement of the parking space number.
1918 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
1919 invalid input and will be assumed to mean that no timeout is desired.
1923 * Added DNS manager support to registrations for peers referencing peer entries.
1924 DNS manager runs in the background which allows DNS lookups to be run asynchronously
1925 as well as periodically updating the IP address. These properties allow for
1926 better performance as well as recovery in the event of an IP change.
1927 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
1928 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
1929 These changes also provide performance improvements for call setup and tear down.
1930 * Added ability to specify registration expiry time on a per registration basis in
1932 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
1934 * Added t38pt_usertpsource option. See sip.conf.sample for details.
1935 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
1936 * 'sip show peers' and 'sip show users' display their entries sorted in
1937 alphabetical order, as opposed to the order they were in, in the config
1939 * Videosupport now supports an additional option, "always", which always sets
1940 up video RTP ports, even on clients that don't support it. This helps with
1941 callfiles and certain transfers to ensure that if two video phones are
1942 connected, they will always share video feeds.
1946 * Existing DNS manager lookups extended to check for SRV records.
1947 * IAX2 encryption support has been improved to support periodic key rotation
1948 within a call for enhanced security. The option "keyrotate" has been
1949 provided to disable this functionality to preserve backwards compatibility
1950 with older versions of IAX2 that do not support key rotation.
1954 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
1955 data tree based on the given <path>.
1956 * New CLI command "data show providers" that will display all the registered
1958 * New CLI command, "config reload <file.conf>" which reloads any module that
1959 references that particular configuration file. Also added "config list"
1960 which shows which configuration files are in use.
1961 * New CLI commands, "pri show version" and "ss7 show version" that will
1962 display which version of libpri and libss7 are being used, respectively.
1963 A new API call was added so trunk will now have to be compiled against
1964 a versions of libpri and libss7 that have them or it will not know that
1965 these libraries exist.
1966 * The commands "core show globals", "core set global" and "core set chanvar" has
1967 been deprecated in favor of the more semanticly correct "dialplan show globals",
1968 "dialplan set chanvar" and "dialplan set global".
1969 * New CLI command "dialplan show chanvar" to list all variables associated
1970 with a given channel.
1974 * Addresses managed by DNS manager now can check to see if there is a DNS
1975 SRV record for a given domain and will use that hostname/port if present.
1977 AMI - The manager (TCP/TLS/HTTP)
1978 --------------------------------
1979 * The Status command now takes an optional list of variables to display
1980 along with channel status.
1981 * The QueueEntry event now also includes the channel's uniqueid
1985 * res_odbc no longer has a limit of 1023 total possible unshared connections,
1986 as some people were running into this limit. This limit has been increased
1991 * The TRANSFER queue log entry now includes the the caller's original
1992 position in the transferred-from queue.
1993 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
1994 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
1995 as well as an explanation about timeout options in general
1996 * Added a new option - C - for forcing the "answered elsewhere" flag on
1997 cancellation of calls in to members of the queue. This is to avoid the
1998 call to a member of a queue having the call listed as a "missed call".
2002 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
2003 adaptive capabilities. What this means in practical terms is that if your
2004 realtime table lacks critical fields, Asterisk will now emit warnings to
2005 that effect. Also, some of the realtime drivers have the ability (if
2006 configured) to automatically add those columns to the table with the
2007 correct type and length.
2011 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
2012 the 'setvar' option to cause a given audio file to be played upon completion
2013 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
2014 Skinny channels only.
2015 * You can now compile Asterisk against the Hoard Memory Allocator, see the
2016 Hoard page on the Asterisk wiki for more information:
2017 https://wiki.asterisk.org/wiki/x/pQBB
2018 * Config file variables may now be appended to, by using the '+=' append
2019 operator. This is most helpful when working with long SQL queries in
2020 func_odbc.conf, as the queries no longer need to be specified on a single
2022 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
2023 which will add a second to the billsec when the ending
2024 time is set, if the number in the microseconds field of the end time is
2025 greater than the number of microseconds in the answer time. This allows
2026 users to count the 'initiated' seconds in their billing records.
2028 ------------------------------------------------------------------------------
2029 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
2030 ------------------------------------------------------------------------------
2032 AMI - The manager (TCP/TLS/HTTP)
2033 --------------------------------
2034 * Manager has undergone a lot of changes, all of them documented
2035 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
2036 * Manager version has changed to 1.1
2037 * Added a new action 'CoreShowChannels' to list currently defined channels
2038 and some information about them.
2039 * Added a new action 'SIPshowregistry' to list SIP registrations.
2040 * Added TLS support for the manager interface and HTTP server
2041 * Added the URI redirect option for the built-in HTTP server
2042 * The output of CallerID in Manager events is now more consistent.
2043 CallerIDNum is used for number and CallerIDName for name.
2044 * Enable https support for builtin web server.
2045 See configs/http.conf.sample for details.
2046 * Added a new action, GetConfigJSON, which can return the contents of an
2047 Asterisk configuration file in JSON format. This is intended to help
2048 improve the performance of AJAX applications using the manager interface
2050 * SIP and IAX manager events now use "ChannelType" in all cases where we
2051 indicate channel driver. Previously, we used a mixture of "Channel"
2052 and "ChannelDriver" headers.
2053 * Added a "Bridge" action which allows you to bridge any two channels that
2054 are currently active on the system.
2055 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
2056 the voicemail users setup.
2057 * Added 'DBDel' and 'DBDelTree' manager commands.
2058 * cdr_manager now reports events via the "cdr" level, separating it from
2059 the very verbose "call" level.
2060 * Manager users are now stored in memory. If you change the manager account
2061 list (delete or add accounts) you need to reload manager.
2062 * Added Masquerade manager event for when a masquerade happens between
2064 * Added "manager reload" command for the CLI
2065 * Lots of commands that only provided information are now allowed under the
2066 Reporting privilege, instead of only under Call or System.
2067 * The IAX* commands now require either System or Reporting privilege, to
2068 mirror the privileges of the SIP* commands.
2069 * Added ability to retrieve list of categories in a config file.
2070 * Added ability to retrieve the content of a particular category.
2071 * Added ability to empty a context.
2072 * Created new action to create a new file.
2073 * Updated delete action to allow deletion by line number with respect to category.
2074 * Added new action insert to add new variable to category at specified line.
2075 * Updated action newcat to allow new category to be inserted in file above another
2077 * Added new event "JitterBufStats" in the IAX2 channel
2078 * Originate now requires the Originate privilege and, if you want to call out
2079 to a subshell, it requires the System privilege, as well. This was done to
2080 enhance manager security.
2081 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
2082 * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
2083 or manager show command Atxfer from the CLI
2084 * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
2085 details or manager show command IAXregistry from the CLI
2089 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
2090 state in the dialplan, as well as creating custom device states that are
2091 controllable from the dialplan.
2092 * Extend CALLERID() function with "pres" and "ton" parameters to
2093 fetch string representation of calling number presentation indicator
2094 and numeric representation of type of calling number value.
2095 * MailboxExists converted to dialplan function
2096 * A new option to Dial() for telling IP phones not to count the call
2097 as "missed" when dial times out and cancels.
2098 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
2099 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
2100 held for any given channel. Also, locks are automatically freed when a
2102 * Added HINT() dialplan function that allows retrieving hint information.
2103 Hints are mappings between extensions and devices for the sake of
2104 determining the state of an extension. This function can retrieve the list
2105 of devices or the name associated with a hint.
2106 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
2108 * Added SYSINFO() dialplan function which allows retrieval of system information
2109 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
2110 the existence of a dialplan target.
2111 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
2112 upper and lower case, respectively.
2113 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
2114 ID for the call (not the Asterisk call ID or unique ID), provided that the
2115 channel driver supports this. For SIP, you get the SIP call-ID for the
2116 bridged channel which you can store in the CDR with a custom field.
2120 * Added CLI permissions, config file: cli_permissions.conf
2121 default is to allow all commands for every local user/group.
2122 Also this new feature added three new CLI commands:
2123 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
2124 - cli reload permissions
2125 - cli show permissions
2126 * New CLI command "core show hint" (usage: core show hint <exten>)
2127 * New CLI command "core show settings"
2128 * Added 'core show channels count' CLI command.
2129 * Added the ability to set the core debug and verbose values on a per-file basis.
2130 * Added 'queue pause member' and 'queue unpause member' CLI commands
2131 * Ability to set process limits ("ulimit") without restarting Asterisk
2132 * Enhanced "agi debug" to print the channel name as a prefix to the debug
2133 output to make debugging on busy systems much easier.
2134 * New CLI commands "dialplan set extenpatternmatching true/false"
2135 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
2136 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
2137 listed in the startup_commands section of cli.conf will get executed.
2138 * Added a CLI command, "devstate change", which allows you to set custom device
2139 states from the func_devstate module that provides the DEVICE_STATE() function
2140 and handling of the "Custom:" devices.
2141 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
2142 sorted into the different possible callbacks, with the number of entries
2143 currently scheduled for each. Gives you a feel for how busy the sip channel
2145 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
2146 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
2147 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
2151 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
2152 option is enabled, Asterisk will watch for a CNG tone in the incoming audio
2153 for a received call. If it is detected, the channel will jump to the
2154 'fax' extension in the dialplan.
2155 * The default SIP useragent= identifier now includes the Asterisk version
2156 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
2157 If set, and the incoming request carries authentication info,
2158 the username to match in the users list is taken from the Digest header
2159 rather than from the From: field. This feature is considered experimental.
2160 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
2161 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
2162 * The "localmask" setting was removed in version 1.2 and the reminder about it
2163 being removed is now also removed.
2164 * A new option "busylevel" for setting a level of calls where asterisk reports
2165 a device as busy, to separate it from call-limit. This value is also added
2166 to the SIP_PEER dialplan function.
2167 * A new realtime family called "sipregs" is now supported to store SIP registration
2168 data. If this family is defined, "sippeers" will be used for configuration and
2169 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
2170 registration data, as before.
2171 * The SIPPEER function have new options for port address, call and pickup groups
2172 * Added support for T.140 realtime text in SIP/RTP
2173 * The "checkmwi" option has been removed from sip.conf, as it is no longer
2174 required due to the restructuring of how MWI is handled. See the descriptions
2175 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
2176 for more information.
2177 * Added rtpdest option to CHANNEL() dialplan function.
2178 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
2179 * SIP now adds a header to the CANCEL if the call was answered by another phone
2180 in the same dial command, or if the new c option in dial() is used.
2181 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
2182 states it is not needed. For phones, however, that do require it the "registertrying" option
2183 has been added so it can be enabled.
2184 * A new option called "callcounter" (global/peer/user level) enables call counters needed
2185 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
2186 used to enable this functionality).
2187 * New settings for timer T1 and timer B on a global level or per device. This makes it
2188 possible to force timeout faster on non-responsive SIP servers. These settings are
2189 considered advanced, so don't use them unless you have a problem.
2190 * Added a dial string option to be able to set the To: header in an INVITE to any
2192 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
2193 the qualify frequency.
2194 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
2195 were not properly torn down due to network or endpoint failures during an established
2197 * Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB
2198 and configs/sip.conf.sample for more information on how it is used.
2199 * Added a new configuration option "authfailureevents" that enables manager events when
2200 a peer can't authenticate properly.
2201 * Added DNS manager support to registrations for peers not referencing a peer entry.
2205 * Added the trunkmaxsize configuration option to chan_iax2.
2206 * Added the srvlookup option to iax.conf
2207 * Added support for OSP. The token is set and retrieved through the CHANNEL()
2210 XMPP Google Talk/Jingle changes
2211 -------------------------------
2212 * Added the bindaddr option to gtalk.conf.
2216 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
2217 * Proper codec support in chan_skinny.
2218 * Added settings for IP and Ethernet QoS requests
2222 * Added separate settings for media QoS in mgcp.conf
2224 Console Channel Driver changes
2225 ------------------------------
2226 * Added experimental support for video send & receive to chan_oss.
2227 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
2230 Phone channel changes (chan_phone)
2231 ----------------------------------
2232 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
2234 H.323 channel Changes
2235 ---------------------
2236 * H323 remote hold notification support added (by NOTIFY message
2237 and/or H.450 supplementary service)
2239 Local channel changes
2240 ---------------------
2241 * The device state functionality in the Local channel driver has been updated
2242 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
2243 to just UNKNOWN if the extension exists.
2244 * Added jitterbuffer support for chan_local. This allows you to use the
2245 generic jitterbuffer on incoming calls going to Asterisk applications.
2246 For example, this would allow you to use a jitterbuffer for an incoming
2247 SIP call to Voicemail by putting a Local channel in the middle. This
2248 feature is enabled by using the 'j' option in the Dial string to the Local
2249 channel in conjunction with the existing 'n' option for local channels.
2250 * A 'b' option has been added which causes chan_local to return the actual channel
2251 that is behind it when queried. This is useful for transfer scenarios as the
2252 actual channel will be transferred, not the Local channel.
2254 Agent channel changes
2255 ----------------------
2256 * The ackcall and endcall options are now supplemented with options acceptdtmf
2257 and enddtmf. These allow for the DTMF keypress to be configurable. The options
2258 default to their old hard-coded values ('#' and '*' respectively) so this should
2259 not break any existing agent installations.
2261 DAHDI channel driver (chan_dahdi) Changes
2262 ----------------------------------------
2263 * SS7 support (via libss7 library)
2264 * In India, some carriers transmit CID via dtmf. Some code has been added
2265 that will handle some situations. The cidstart=polarity_IN choice has been added for
2266 those carriers that transmit CID via dtmf after a polarity change.
2267 * CID matching information is now shown when doing 'dialplan show'.
2268 * Added dahdi show version CLI command.
2269 * Added setvar support to chan_dahdi.conf channel entries.
2270 * Added two new options: mwimonitor and mwimonitornotify. These options allow
2271 you to enable MWI monitoring on FXO lines. When the MWI state changes,
2272 the script specified in the mwimonitornotify option is executed. An internal
2273 event indicating the new state of the mailbox is also generated, so that
2274 the normal MWI facilities in Asterisk work as usual.
2275 * Added signalling type 'auto', which attempts to use the same signalling type
2276 for a channel as configured in DAHDI. This is primarily designed for analog
2277 ports, but will also work for digital ports that are configured for FXS or FXO
2278 signalling types. This mode is also the default now, so if your chan_dahdi.conf
2279 does not specify signalling for a channel (which is unlikely as the sample
2280 configuration file has always recommended specifying it for every channel) then
2281 the 'auto' mode will be used for that channel if possible.
2282 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
2283 state for a channel; also ensured that the DNDState Manager event is
2284 emitted no matter how the DND state is set or cleared.
2288 * Added a new channel driver, chan_unistim. See the Asterisk wiki at
2289 https://wiki.asterisk.org/wiki/x/vgsiAQ and configs/unistim.conf.sample
2290 for details. This new channel driver allows you to use Nortel i2002,
2291 i2004, and i2050 phones with Asterisk.
2292 * Added a new channel driver, chan_console, which uses portaudio as a cross
2293 platform audio interface. It was written as a channel driver that would
2294 work with Mac CoreAudio, but portaudio supports a number of other audio
2295 interfaces, as well. Note that this channel driver requires v19 or higher
2296 of portaudio; older versions have a different API.
2300 * Added the ability to specify arguments to the Dial application when using
2301 the DUNDi switch in the dialplan.
2302 * Added the ability to set weights for responses dynamically. This can be
2303 done using a global variable or a dialplan function. Using the SHELL()
2304 function would allow you to have an external script set the weight for
2306 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
2307 functions will allow you to initiate a DUNDi query from the dialplan,
2308 find out how many results there are, and access each one.
2309 * Added the ability to specifiy a port for a dundi peer.
2313 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
2314 functions will allow you to initiate an ENUM lookup from the dialplan,
2315 and Asterisk will cache the results. ENUMRESULT can be used to access
2316 the results without doing multiple DNS queries.
2320 * Added the ability to customize which sound files are used for some of the
2321 prompts within the Voicemail application by changing them in voicemail.conf
2322 * Added the ability for the "voicemail show users" CLI command to show users
2323 configured by the dynamic realtime configuration method.
2324 * MWI (Message Waiting Indication) handling has been significantly
2325 restructured internally to Asterisk. It is now totally event based
2326 instead of polling based. The voicemail application will notify other
2327 modules that have subscribed to MWI events when something in the mailbox
2329 This also means that if any other entity outside of Asterisk is changing
2330 the contents of mailboxes, then the voicemail application still needs to
2331 poll for changes. Examples of situations that would require this option
2332 are web interfaces to voicemail or an email client in the case of using
2333 IMAP storage. So, two new options have been added to voicemail.conf
2334 to account for this: "pollmailboxes" and "pollfreq". See the sample
2335 configuration file for details.
2336 * Added "tw" language support
2337 * Added support for storage of greetings using an IMAP server
2338 * Added ability to customize forward, reverse, stop, and pause keys for message playback
2339 * SMDI is now enabled in voicemail using the smdienable option.
2340 * A "lockmode" option has been added to asterisk.conf to configure the file
2341 locking method used for voicemail, and potentially other things in the
2342 future. The default is the old behavior, lockfile. However, there is a
2343 new method, "flock", that uses a different method for situations where the
2344 lockfile will not work, such as on SMB/CIFS mounts.
2345 * Added the ability to backup deleted messages, to ease recovery in the case
2346 that a user accidentally deletes a message, and discovers that they need it.
2347 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
2348 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
2349 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
2350 voicemail boxes. The SMDI interface can also poll for MWI changes when some
2351 outside entity is modifying the state of the mailbox (such as IMAP storage or
2352 a web interface of some kind).
2353 * Added the support for marking messages as "urgent." There are two methods to accomplish
2354 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
2355 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
2356 the message as urgent after he has recorded a voicemail by following the voice instructions.
2357 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
2362 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
2363 used across multiple queues.
2364 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
2365 setqueueentryvar options for each queue, see queues.conf.sample for details.
2366 * Added keepstats option to queues.conf which will keep queue
2367 statistics during a reload.
2368 * setinterfacevar option in queues.conf also now sets a variable
2369 called MEMBERNAME which contains the member's name.
2370 * Added 'Strategy' field to manager event QueueParams which represents
2371 the queue strategy in use.
2372 * Added option to run macro when a queue member is connected to a caller,
2373 see queues.conf.sample for details.
2374 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
2375 does not count paused queue members as unavailable.
2376 * Added min-announce-frequency option to queues.conf which allows you to control the
2377 minimum amount of time between queue announcements for use when the caller's queue
2378 position changes frequently.
2379 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
2381 * Added ability for non-realtime queues to have realtime members
2382 * Added the "linear" strategy to queues.
2383 * Added the "wrandom" strategy to queues.
2384 * Added new channel variable QUEUE_MIN_PENALTY
2385 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
2386 rules in queuerules.conf. See configs/queuerules.conf.sample for details
2387 * Added a new parameter for member definition, called state_interface. This may be
2388 used so that a member may be called via one interface but have a different interface's
2389 device state reported.
2390 * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
2391 "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
2392 "manager show command QueueReset."
2393 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
2394 specified by the periodic-announce option, then one will be chosen randomly when it is time
2395 to play a periodic announcment
2396 * New configuration options: announce-position now takes two more values in addition to "yes" and
2397 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
2398 announce-position-limit. By setting announce-position to "limit" callers will only have their
2399 position announced if their position is less than what is specified by announce-position-limit.
2400 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
2401 will be told that their are more than announce-position-limit callers waiting.
2402 * Two new queue log events have been added. An ADDMEMBER event will be logged
2403 when a realtime queue member is added and a REMOVEMEMBER event will be logged
2404 when a realtime queue member is removed. Since there is no calling channel associated
2405 with these events, the string "REALTIME" is placed where the channel's unique id
2406 is typically placed.
2407 * The configuration method for the "joinempty" and "leavewhenempty" options has
2408 changed to a comma-separated list of methods of determining member availability
2409 instead of vague terms such as "yes," "loose," "no," and "strict." These old four
2410 values are still accepted for backwards-compatibility, though.
2411 * The average talktime is now calculated on queues. This information is reported via the
2412 CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
2413 and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
2418 * The 'o' option to provide an optimization has been removed and its functionality
2419 has been enabled by default.
2420 * When a conference is created, the UNIQUEID of the channel that caused it to be
2421 created is stored. Then, every channel that joins the conference will have the
2422 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
2423 callers that come and go from long standing conferences.
2424 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
2425 except it does operations on a channel by name, instead of number in a conference.
2426 This is a very useful feature in combination with the 'X' option to ChanSpy.
2427 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
2429 * Added new RealTime functionality to provide support for scheduled conferencing.
2430 This includes optional messages to the caller if they attempt to join before
2431 the schedule start time, or to allow the caller to join the conference early.
2432 Also included is optional support for limiting the number of callers per
2433 RealTime conference.
2434 * Added the S() and L() options to the MeetMe application. These are pretty
2435 much identical to the S() and L() options to Dial(). They let you set
2436 timeouts for the conference, as well as have warning sounds played to
2437 let the caller know how much time is left, and when it is running out.
2438 * Added the ability to do "meetme concise" with the "meetme" CLI command.
2439 This extends the concise capabilities of this CLI command to include
2440 listing all conferences, instead of an addition to the other sub commands
2441 for the "meetme" command.
2442 * Added the ability to specify the music on hold class used to play into the
2443 conference when there is only one member and the M option is used.
2444 * Added MEETME_INFO dialplan function which provides a way to query
2445 various properties of a Meetme conference.
2446 * Added new admin features: *81: Roll call, *82: eject all, *83: mute all,
2447 and *84: record in-conf
2449 Other Dialplan Application Changes
2450 ----------------------------------
2451 * Argument support for Gosub application
2452 * From the to-do lists: straighten out the app timeout args:
2453 Wait() app now really does 0.3 seconds- was truncating arg to an int.
2454 WaitExten() same as Wait().
2455 Congestion() - Now takes floating pt. argument.
2456 Busy() - now takes floating pt. argument.
2457 Read() - timeout now can be floating pt.
2458 WaitForRing() now takes floating pt timeout arg.
2459 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
2460 * Added 's' option to Page application.
2461 * Added an optional timeout argument to the Page application.
2462 * Added 'E', 'V', and 'P' commands to ExternalIVR.
2463 * Added 'o' and 'X' options to Chanspy.
2464 * Added a new dialplan application, Bridge, which allows you to bridge the
2465 calling channel to any other active channel on the system.
2466 * Added the ability to specify a music on hold class to play instead of ringing
2467 for the SLATrunk application.
2468 * The Read application no longer exits the dialplan on error. Instead, it sets
2469 READSTATUS to ERROR, which you can catch and handle separately.
2470 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
2471 of asking for verification of each name, one at a time.
2472 * Privacy() no longer uses privacy.conf, as all options are specifyable as
2473 direct options to the app.
2474 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
2476 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
2477 * The ChannelRedirect application no longer exits the dialplan if the given channel
2478 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
2479 or NOCHANNEL if the given channel was not found.
2480 * The silencethreshold setting that was previously configurable in multiple
2481 applications is now settable globally via dsp.conf.
2483 Music On Hold Changes
2484 ---------------------
2485 * A new option, "digit", has been added for music on hold classes in
2486 musiconhold.conf. If this is set for a music on hold class, a caller
2487 listening to music on hold can press this digit to switch to listening
2488 to this music on hold class.
2489 * Support for realtime music on hold has been added.
2490 * In conjunction with the realtime music on hold, a general section has
2491 been added to musiconhold.conf, its sole variable is cachertclasses. If this
2492 is set, then music on hold classes found in realtime will be cached in memory.
2496 * AEL upgraded to use the Gosub with Arguments instead
2497 of Macro application, to hopefully reduce the problems
2498 seen with the artificially low stack ceiling that
2499 Macro bumps into. Macros can only call other Macros
2500 to a depth of 7. Tests run using gosub, show depths
2501 limited only by virtual memory. A small test demonstrated
2502 recursive call depths of 100,000 without problems.
2503 -- in addition to this, all apps that allowed a macro
2504 to be called, as in Dial, queues, etc, are now allowing
2505 a gosub call in similar fashion.
2506 * AEL now generates LOCAL(argname) declarations when it
2507 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
2508 etc. That makes the arguments local in scope. The user
2509 can define their own local variables in macros, now,
2510 by saying "local myvar=someval;" or using Set() in this
2511 fashion: Set(LOCAL(myvar)=someval); ("local" is now
2513 * utils/conf2ael introduced. Will convert an extensions.conf
2514 file into extensions.ael. Very crude and unfinished, but
2515 will be improved as time goes by. Should be useful for a
2516 first pass at conversion.
2517 * aelparse will now read extensions.conf to see if a referenced
2518 macro or context is there before issueing a warning.
2519 * AEL parser sets a local channel variable ~~EXTEN~~, to
2520 preserve the value of ${EXTEN} thru switch statements.
2521 * New operator in $[...] expressions: the ~~ operator serves
2522 as a concatenation operator. AT THE MOMENT, it is really only
2523 necessary and useful in AEL, especially in if() expressions.
2524 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
2525 any enclosing double-quotes, and evaluate to the value of a
2526 concatenated with the value of b. For example if a is set to
2527 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
2528 evaluate to xyzabc .
2531 Call Features (res_features) Changes
2532 ------------------------------------
2533 * Added the parkedcalltransfers option to features.conf
2534 * Added parkedcallparking option to control one touch parking w/ parking
2536 * Added parkedcallhangup option to control disconnect feature w/ parking
2538 * Added parkedcallrecording option to control one-touch record w/ parking
2540 * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
2541 parkedcalltransfers option support for multiple parking lots.
2542 * Added BRIDGE_FEATURES variable to set available features for a channel
2543 * The built-in method for doing attended transfers has been updated to
2544 include some new options that allow you to have the transferee sent
2545 back to the person that did the transfer if the transfer is not successful.
2546 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
2547 in features.conf.sample.
2548 * Added support for configuring named groups of custom call features in
2549 features.conf. This means that features can be written a single time, and
2550 then mapped into groups of features for different key mappings or easier
2552 * Updated the ParkedCall application to allow you to not specify a parking
2553 extension. If you don't specify a parking space to pick up, it will grab
2554 the first one available.
2555 * Added cli command 'features reload' to reload call features from features.conf
2556 * Moved into core asterisk binary.
2557 * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
2558 * Added the ability for custom parking lots to be configured with their own
2559 parking extension with the parkext option.
2561 Language Support Changes
2562 ------------------------
2563 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
2564 * Added support for the Hungarian language for saying numbers, dates, and times.
2568 * Added SPEECH commands for speech recognition. A complete listing can be found
2570 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
2571 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
2572 does not behave as expected; the native command needs to be used, instead.
2573 * Added the ability to perform SRV lookups on fast AGI calls. To use this
2574 feature, simply use hagi: instead of agi: as the protocol portion
2575 of the URI parameter to the AGI function call in your dial plan. Also note
2576 that specifying a port number in the AGI URI will disable SRV lookups,
2577 even if you use the hagi: protocol.
2578 * No longer support MSG_OOB flag on HANGUP.
2582 * Added rotatestrategy option to logger.conf, along with two new options:
2583 "timestamp" which will use the time to name the logger files instead of
2584 sequence number; and "rotate", which rotates the names of the log files,
2585 similar to the way syslog rotates files.
2586 * Added exec_after_rotate option to logger.conf, which allows a system
2587 command to be run after rotation. This is primarily useful with
2588 rotatestrategy=rotate, to allow a limit on the number of log files kept
2589 and to ensure that the oldest log file gets deleted.
2590 * Added realtime support for the queue log
2594 * The cdr_manager module has a [mappings] feature, like cdr_custom,
2595 to add fields to the manager event from the CDR variables.
2596 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
2597 backend database CDR table. Specifically, additional, non-standard
2598 columns are supported, merely by setting the corresponding CDR variable in
2599 your dialplan. In addition, you may alias any column to another name (for
2600 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
2601 simply "alias src => ANI" in the configuration file). Records may be
2602 posted to more than one backend, simply by specifying multiple categories
2603 in the configuration file. And finally, you may filter which CDRs get
2604 posted to each backend, by specifying a filter (which the record must
2605 match) for the particular category. Filters are additive (meaning all
2606 rules must match to post that CDR).
2607 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
2608 module. Specifically, you may add additional columns into the table and
2609 they will be set, if you set the corresponding CDR variable name. Also,
2610 if you omit columns in your database table, they will be silently skipped
2611 (but a record will still be inserted, based on what columns remain). Note
2612 that the other two features from cdr_adaptive_odbc (alias and filter) are
2613 not currently supported.
2614 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
2615 has been disabled using the NoCDR application.
2617 Miscellaneous New Modules
2618 -------------------------
2619 * Added a new CDR module, cdr_sqlite3_custom.
2620 * Added a new realtime configuration module, res_config_sqlite
2621 * Added a new codec translation module, codec_resample, which re-samples
2622 signed linear audio between 8 kHz and 16 kHz to help support wideband
2624 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
2625 based on configuration templates that use Asterisk dialplan function and
2626 variable substitution. It should be possible to create phone profiles and
2627 templates that work for the majority of phones provisioned over http. It
2628 is currently only intended to provision a single user account per phone.
2629 An example profile and set of templates for Polycom phones is provided.
2630 NOTE: Polycom firmware is not included, but should be placed in
2631 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
2632 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
2633 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
2634 provided; there is a JACK() application, and a JACK_HOOK() function. Both
2635 interfaces create an input and output JACK port. The application makes
2636 these ports the endpoint of the call. The audio coming from the channel
2637 goes out the output port and whatever comes back in on the input port is
2638 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
2639 audiohook on the channel. This lets you run the audio coming from a
2640 channel through JACK, and whatever comes back in is what gets forwarded
2641 on as the channel's audio. This is very useful for building custom
2642 vocoders or doing recording or analysis of the channel's audio in another
2644 * Added a new module, res_config_curl, which permits using a HTTP POST url
2645 to retrieve, create, update, and delete realtime information from a remote
2646 web server. Note that this module requires func_curl.so to be loaded for
2647 backend functionality.
2648 * Added a new module, res_config_ldap, which permits the use of an LDAP
2649 server for realtime data access.
2650 * Added support for writing and running your dialplan in lua using the pbx_lua
2651 module. See configs/extensions.lua.sample for examples of how to do this.
2655 * Ability to use libcap to set high ToS bits when non-root
2656 on Linux. If configure is unable to find libcap then you
2657 can use --with-cap to specify the path.
2658 * Added maxfiles option to options section of asterisk.conf which allows you to specify
2659 what Asterisk should set as the maximum number of open files when it loads.
2660 * Added the jittertargetextra configuration option.
2661 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
2662 configuration files for the IP channel drivers. The new option is "cos".
2663 This information is also documented on the Asterisk wiki at
2664 https://wiki.asterisk.org/wiki/x/EYBG
2665 * When originating a call using AMI or pbx_spool that fails the reason for failure
2666 will now be available in the failed extension using the REASON dialplan variable.
2667 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
2668 It allows you to configure a prefix for auto-monitor recordings.
2669 * A new extension pattern matching algorithm, based on a trie, is introduced
2670 here, that could noticeably speed up mid-sized to large dialplans.
2671 It is NOT used by default, as duplicating the behaviour of the old pattern
2672 matcher is still under development. A config file option, in extensions.conf,
2673 in the [general] section, called "extenpatternmatchingnew", is by default
2674 set to false; setting that to true will force the use of the new algorithm.
2675 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
2676 be used to switch the algorithms at run time.
2677 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
2678 specifying which socket to use to connect to the running Asterisk daemon
2680 * Performance enhancements to the sched facility, which is used in
2681 the channel drivers, etc. Added hashtabs and doubly-linked lists
2682 to speed up deletion; start at the beginning or end of list to
2684 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
2685 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
2686 Added regression tests to the tests/ dir, also.
2687 * Added a refcount trace feature to astobj2 for those trying to balance
2688 object creation, deletion; work, play; space and time. See the
2689 notes in astobj2.h. Also, see utils/refcounter as well, as a
2690 quick way to find unbalanced refcounts in what could be a sea
2691 of objects that were balanced.
2692 * Added logging to 'make update' command. See update.log
2693 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
2694 do not come from the remote party.
2695 * Added the 'n' option to the SpeechBackground application to tell it to not
2696 answer the channel if it has not already been answered.
2697 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
2698 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
2700 * iLBC source code no longer included (see UPGRADE.txt for details)
2701 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
2702 deadlock is detected, a backtrace of the stack which led to the lock calls
2703 will be output to the CLI.
2704 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
2705 the "core show locks" CLI command will give lock information output as well
2706 as a backtrace of the stack which led to the lock calls.
2707 * users.conf now sports an optional alternateexts property, which permits
2708 allocation of additional extensions which will reach the specified user.
2709 * A new option for the configure script, --enable-internal-poll, has been added
2710 for use with systems which may have a buggy implementation of the poll system
2711 call. If you notice odd behavior such as the CLI being unresponsive on remote
2712 consoles, you may want to try using this option. This option is enabled by default
2713 on Darwin systems since it is known that the Darwin poll() implementation has
2717 --------------------
2718 * In addition to timing from DAHDI, there is a new timing module called
2719 res_timing_timerfd. In order to use this, you must be running Linux with
2720 a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
2721 script will be able to tell if you have the requirements. From menuselect, select
2722 res_timing_timerfd from the Resource Modules menu.
projects / asterisk/asterisk.git / blob