1 ======================================================================
3 === This file documents the new and/or enhanced functionality added in
4 === the Asterisk versions listed below. This file does NOT include
5 === changes in behavior that would not be backwards compatible with
6 === previous versions; for that information see the UPGRADE.txt file
7 === and the other UPGRADE files for older releases.
9 ======================================================================
11 ------------------------------------------------------------------------------
12 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
13 ------------------------------------------------------------------------------
17 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
18 Snom phones use this for call pickup of extensions that the phone is
20 * Added support for subscribing to a voice mailbox on a remote server and
21 making the new/old message count available to local devices.
22 * Added support for setting the domain in the URI for caller of an
23 outbound call by using the SIPFROMDOMAIN channel variable.
24 * Added a new configuration option "remotesecret" for authentication to
25 remote services. For backwards compatibility, "secret" still has the
26 same function as before, but now you can configure both a remote secret and a
27 local secret for mutual authentication.
28 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
29 option is enabled, a SIP channel will go to the fax extension (if it exists)
30 after T38 is negotiated. This option is disabled by default.
31 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
32 the sound will be played to the target of an attended transfer
33 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
34 finer control over how many peers Asterisk will qualify and the gap between them
35 when all peers need to be qualified at the same time.
36 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
37 (either globally or for a specific peer), chan_sip will treat any SDP data
38 it receives as new data and update the media stream accordingly. By
39 default, Asterisk will only modify the media stream if the SDP session
40 version received is different from the current SDP session version. This
41 option is required to interoperate with devices that have non-standard SDP
42 session version implementations (observed with Microsoft OCS). This option
43 is disabled by default.
44 * The parsing of register => lines in sip.conf has been modified to allow a port
45 to be present in the "user" portion. Please see the sip.conf.sample file for more
47 * Added support for subscribing to MWI on a remote server and making the status available
48 as a mailbox. Please see the sip.conf.sample file for more information.
49 * Added a function to remove SIP headers added in the dialplan before the
50 first INVITE is generated - SIPRemoveHeader()
51 * Channel variables set with setvar= in a device configuration is now
52 set both for inbound and outbound calls.
53 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
57 * Added immediate option to iax.conf
58 * Added forceencryption option to iax.conf
59 * Added Encryption and Trunk status to manager command "iaxpeers"
63 * The configuration file now holds separate sections for devices and lines.
64 Please have a look at configs/skinny.conf.sample and change your skinny.conf
69 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
70 support for LibOpenR2. http://www.libopenr2.org/
71 * The UK option waitfordialtone has been added for use with BT analog
73 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
74 is used in conjunction with the 'faxdetect' configuration option. When
75 'faxbuffers' is used and fax tones are detected, the channel will dynamically
76 switch to the configured faxbuffers policy. For example, to use 6 buffers
77 and a 'full' buffer policy for a fax transmission, add:
79 The faxbuffers configuration will be in affect until the call is torn down.
83 * Added a new dialplan function, CURLOPT, which permits setting various
84 options that may be useful with the CURL dialplan function, such as
85 cookies, proxies, connection timeouts, passwords, etc.
86 * Permit the syntax and synopsis fields of the corresponding dialplan
87 functions to be individually set from func_odbc.conf.
88 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
89 * func_odbc now may specify an insert query to execute, when the write query
90 affects 0 rows (usually indicating that no such row exists).
91 * Added a new dialplan function, LISTFILTER, which permits removing elements
92 from a set list, by name. Uses the same general syntax as the existing CUT
93 and FIELDQTY dialplan functions, which also manage lists.
94 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
95 obtaining realtime data from the dialplan.
96 * Added LOCAL_PEEK, which I have no idea how to use, but Leif Madsen wanted it.
97 Russell says it's, like, a scope resolution function for LOCAL variables.
98 Totally. Hopefully, that means more to you than it does to me.
99 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
100 of "core show function AUDIOHOOK_INHERIT" from the CLI
101 * Added AES_ENCRYPT. For information on its use, please see the output
102 of "core show function AES_ENCRYPT" from the CLI
103 * Added AES_DECRYPT. For information on its use, please see the output
104 of "core show function AES_DECRYPT" from the CLI
105 * func_odbc now supports database transactions across multiple queries.
109 * DAHDISendCallreroutingFacility parameters are now comma-separated,
110 instead of the old pipe.
111 * Scheduled meetme conferences may now have their end times extended by
113 * app_authenticate now gives the ability to select a prompt other than
115 * app_directory now pays attention to the searchcontexts setting in
116 voicemail.conf and will look through all contexts, if no context is
117 specified in the initial argument.
118 * A new application, Originate, has been introduced, that allows asynchronous
119 call origination from the dialplan.
120 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
121 in addition to the setting in the "general" context.
122 * Added ConfBridge dialplan application which does conference bridges without
123 DAHDI. For information on its use, please see the output of
124 "core show application ConfBridge" from the CLI.
128 * The Asterisk CLI has a new command, "channel redirect", which is similar in
129 operation to the AMI Redirect action.
130 * res_jabber: autoprune has been disabled by default, to avoid misconfiguration
131 that would end up being interpreted as a bug once Asterisk started removing
132 the contacts from a user list.
133 * extensions.conf now allows you to use keyword "same" to define an extension
134 without actually specifying an extension. It uses exactly the same pattern
135 as previously used on the last "exten" line. For example:
136 exten => 123,1,NoOp(something)
137 same => n,SomethingElse()
138 * musiconhold.conf classes of type 'files' can now use relative directory paths,
139 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
140 * All deprecated CLI commands are removed from the sourcecode. They are now handled
141 by the new clialiases module. See cli_aliases.conf.sample file.
142 * Times within timespecs are now accurate down to the minute. This is a change
143 from historical Asterisk, which only provided timespecs rounded to the nearest
144 even (read: evenly divisible by 2) minute mark.
145 * The realtime switch now supports an option flag, 'p', which disables searches for
147 * In addition to a time range and date range, timespecs now accept a 5th optional
148 argument, timezone. This allows you to perform time checks on alternate
149 timezones, especially if those daylight savings time ranges vary from your
150 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
152 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
153 give you the correct output for an asterisk box behind nat. It will give you the
154 externhost and localnet settings.
155 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
156 can connect calls in passthrough mode, as well as record and play back files.
157 * Successful and unsuccessful call pickup can now be alerted through sounds, by
158 using pickupsound and pickupfailsound in features.conf.
159 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
160 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
161 instead of the /var/run/asterisk.pid where it used to be. This will make
162 installs as non-root easier to manage.
164 Asterisk Manager Interface
165 --------------------------
166 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
167 a non-empty value) in your request. If you do this, any pending AMI events will
168 *not* be included in the response to your request as they would normally, but
169 will be left in the event queue for the next request you make to retrieve. For
170 some applications, this will allow you to guarantee that you will only see
171 events in responses to 'WaitEvent' actions, and can better know when to expect them.
172 To know whether the Asterisk server supports this header or not, your client can
173 inspect the first response back from the server to see if it includes this header:
175 Pragma: SuppressEvents
177 If this is included, the server supports event suppression.
179 * Added 4 new Actions to list skinny device(s) and line(s)
185 ------------------------------------------------------------------------------
186 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
187 ------------------------------------------------------------------------------
189 Device State Handling
190 ---------------------
191 * The event infrastructure in Asterisk got another big update to help support
192 distributed events. It currently supports distributed device state and
193 distributed Voicemail MWI (Message Waiting Indication). A new module has
194 been merged, res_ais, which facilitates communicating events between servers.
195 It uses the SAForum AIS (Service Availability Forum Application Interface
196 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
197 a cluster of Asterisk servers, and to share events between them. For more
198 information on setting this up, see doc/distributed_devstate.txt.
202 * Added a new dialplan function, AST_CONFIG(), which allows you to access
203 variables from an Asterisk configuration file.
204 * The JACK_HOOK function now has a c() option to supply a custom client name.
205 * Added two new dialplan functions from libspeex for audio gain control and
206 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
207 rx directions of a channel from the dialplan.
208 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
209 based on other parameters. The default is still to search based on the
210 forwarding station ID. However, there are new options that allow you to search
211 based on the message desk terminal ID, or the message desk number.
212 * TIMEOUT() has been modified to be accurate down to the millisecond.
213 * ENUM*() functions now include the following new options:
214 - 'u' returns the full URI and does not strip off the URI-scheme.
215 - 's' triggers ISN specific rewriting
216 - 'i' looks for branches into an Infrastructure ENUM tree
217 - 'd' for a direct DNS lookup without any flipping of digits.
218 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
219 * CHANNEL() now has options for the maximum, minimum, and standard or normal
220 deviation of jitter, rtt, and loss for a call using chan_sip.
222 DAHDI channel driver (chan_dahdi) Changes
223 ----------------------------------------
224 * Channels can now be configured using named sections in chan_dahdi.conf, just
225 like other channel drivers, including the use of templates.
226 * The default for pridialplan has changed from 'national' to 'unknown'.
230 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
231 to something that matches the pattern a hint will be created using the contents
232 and variables evaluated.
233 * Dialplan matching has been extended to allow an extension to return to the
234 PBX core to wait for more digits. This is done by using the new dialplan
235 application called "Incomplete". This will permit a whole new level of
236 extension control, by giving the administrator more control over early
237 matches employing one of the short-circuit pattern match operators. Note
238 that custom applications can trigger this same behavior by returning the
239 special value AST_PBX_INCOMPLETE.
243 * Directory now permits both first and last names to be matched at the same
244 time. In addition, the number of digits to enter of the name can be set in
245 the arguments to Directory; previously, you could enter only 3, regardless
246 of how many names are in your company. For large companies, this should be
248 * Voicemail now permits a mailbox setting to wrap around from first to last
249 messages, if the "messagewrap" option is set to a true value.
250 * Voicemail now permits an external script to be run, for password validation.
251 The script should output "VALID" or "INVALID" on stdout, depending upon the
252 wish to validate or invalidate the password given. Arguments are:
253 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
255 * Dial has a new option: F(context^extension^pri), which permits a callee to
256 continue in the dialplan, at the specified label, if the caller hangs up.
257 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
258 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
259 * The Jack application now has a c() option to supply a custom client name.
260 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
261 like the pre-existing whisper mode, except that the spy can also talk to the
262 participant on the bridged channel as well.
263 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
264 to be spoken instead of the channel name or number. For more information on the
265 use of this option, issue the command "core show application ChanSpy" from the
267 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
268 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
269 words, if using the 'd' option, it is not possible to enter a number to append to
270 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
271 change to whisper mode, and pressing 6 will change to barge mode.
272 * ExternalIVR now takes several options that affect the way it performs, as
273 well as having several new commands. Please see doc/externalivr.txt for the
274 complete documentation.
275 * Added ability to communicate over a TCP socket instead of forking a child process for the
276 ExternalIVR application.
277 * ChanIsAvail has a new option, 'a', which will return all available channels instead
278 of just the first one if you give the function more then one channel to check.
279 * PrivacyManager now takes an option where you can specify a context where the
280 given number will be matched. This way you have more control over who is allowed
281 and it stops the people who blindly enter 10 digits.
282 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
283 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
284 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
285 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
286 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
287 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
288 * The Dial() application no longer copies the language used by the caller to the callee's
289 channel. If you desire for the caller's channel's language to be used for file playback
290 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
291 * SendImage() no longer hangs up the channel on error; instead, it sets the
292 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
293 'UNSUPPORTED'. This change makes SendImage() more consistent with other
295 * Park has a new option, 's', which silences the announcement of the parking space number.
296 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
297 invalid input and will be assumed to mean that no timeout is desired.
301 * Added DNS manager support to registrations for peers referencing peer entries.
302 DNS manager runs in the background which allows DNS lookups to be run asynchronously
303 as well as periodically updating the IP address. These properties allow for
304 better performance as well as recovery in the event of an IP change.
305 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
306 load/reload of large numbers of peers/users by ~40x (for large lists of peers.
307 Initially, we saw 4x improvement in call setup/destruction, but at the time
308 of merging, this gain has disappeared; further research will be done to try
309 and restore this performance improvement. Astobj2 refcounting is now used
310 for users, peers, and dialogs. Users are encouraged to assist in regression
311 testing and problem reporting!
312 * Added ability to specify registration expiry time on a per registration basis in
314 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
316 * Added t38pt_usertpsource option. See sip.conf.sample for details.
317 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
318 * 'sip show peers' and 'sip show users' display their entries sorted in
319 alphabetical order, as opposed to the order they were in, in the config
321 * Videosupport now supports an additional option, "always", which always sets
322 up video RTP ports, even on clients that don't support it. This helps with
323 callfiles and certain transfers to ensure that if two video phones are
324 connected, they will always share video feeds.
328 * Existing DNS manager lookups extended to check for SRV records.
329 * IAX2 encryption support has been improved to support periodic key rotation
330 within a call for enhanced security. The option "keyrotate" has been
331 provided to disable this functionality to preserve backwards compatibility
332 with older versions of IAX2 that do not support key rotation.
336 * New CLI command, "config reload <file.conf>" which reloads any module that
337 references that particular configuration file. Also added "config list"
338 which shows which configuration files are in use.
339 * New CLI commands, "pri show version" and "ss7 show version" that will
340 display which version of libpri and libss7 are being used, respectively.
341 A new API call was added so trunk will now have to be compiled against
342 a versions of libpri and libss7 that have them or it will not know that
343 these libraries exist.
344 * The commands "core show globals", "core set global" and "core set chanvar" has
345 been deprecated in favor of the more semanticly correct "dialplan show globals",
346 "dialplan set chanvar" and "dialplan set global".
347 * New CLI command "dialplan show chanvar" to list all variables associated
348 with a given channel.
352 * Addresses managed by DNS manager now can check to see if there is a DNS
353 SRV record for a given domain and will use that hostname/port if present.
355 AMI - The manager (TCP/TLS/HTTP)
356 --------------------------------
357 * The Status command now takes an optional list of variables to display
358 along with channel status.
359 * The QueueEntry event now also includes the channel's uniqueid
363 * res_odbc no longer has a limit of 1023 total possible unshared connections,
364 as some people were running into this limit. This limit has been increased
369 * The TRANSFER queue log entry now includes the the caller's original
370 position in the transferred-from queue.
371 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
372 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
373 as well as an explanation about timeout options in general
374 * Added a new option - C - for forcing the "answered elsewhere" flag on
375 cancellation of calls in to members of the queue. This is to avoid the
376 call to a member of a queue having the call listed as a "missed call".
380 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
381 adaptive capabilities. What this means in practical terms is that if your
382 realtime table lacks critical fields, Asterisk will now emit warnings to
383 that effect. Also, some of the realtime drivers have the ability (if
384 configured) to automatically add those columns to the table with the
385 correct type and length.
389 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
390 the 'setvar' option to cause a given audio file to be played upon completion
391 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
392 Skinny channels only.
393 * You can now compile Asterisk against the Hoard Memory Allocator, see doc/hoard.txt
394 for more information.
395 * Config file variables may now be appended to, by using the '+=' append
396 operator. This is most helpful when working with long SQL queries in
397 func_odbc.conf, as the queries no longer need to be specified on a single
399 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
400 which will add a second to the billsec when the ending
401 time is set, if the number in the microseconds field of the end time is
402 greater than the number of microseconds in the answer time. This allows
403 users to count the 'initiated' seconds in their billing records.
405 ------------------------------------------------------------------------------
406 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
407 ------------------------------------------------------------------------------
409 AMI - The manager (TCP/TLS/HTTP)
410 --------------------------------
411 * Manager has undergone a lot of changes, all of them documented
412 in doc/manager_1_1.txt
413 * Manager version has changed to 1.1
414 * Added a new action 'CoreShowChannels' to list currently defined channels
415 and some information about them.
416 * Added a new action 'SIPshowregistry' to list SIP registrations.
417 * Added TLS support for the manager interface and HTTP server
418 * Added the URI redirect option for the built-in HTTP server
419 * The output of CallerID in Manager events is now more consistent.
420 CallerIDNum is used for number and CallerIDName for name.
421 * Enable https support for builtin web server.
422 See configs/http.conf.sample for details.
423 * Added a new action, GetConfigJSON, which can return the contents of an
424 Asterisk configuration file in JSON format. This is intended to help
425 improve the performance of AJAX applications using the manager interface
427 * SIP and IAX manager events now use "ChannelType" in all cases where we
428 indicate channel driver. Previously, we used a mixture of "Channel"
429 and "ChannelDriver" headers.
430 * Added a "Bridge" action which allows you to bridge any two channels that
431 are currently active on the system.
432 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
433 the voicemail users setup.
434 * Added 'DBDel' and 'DBDelTree' manager commands.
435 * cdr_manager now reports events via the "cdr" level, separating it from
436 the very verbose "call" level.
437 * Manager users are now stored in memory. If you change the manager account
438 list (delete or add accounts) you need to reload manager.
439 * Added Masquerade manager event for when a masquerade happens between
441 * Added "manager reload" command for the CLI
442 * Lots of commands that only provided information are now allowed under the
443 Reporting privilege, instead of only under Call or System.
444 * The IAX* commands now require either System or Reporting privilege, to
445 mirror the privileges of the SIP* commands.
446 * Added ability to retrieve list of categories in a config file.
447 * Added ability to retrieve the content of a particular category.
448 * Added ability to empty a context.
449 * Created new action to create a new file.
450 * Updated delete action to allow deletion by line number with respect to category.
451 * Added new action insert to add new variable to category at specified line.
452 * Updated action newcat to allow new category to be inserted in file above another
454 * Added new event "JitterBufStats" in the IAX2 channel
455 * Originate now requires the Originate privilege and, if you want to call out
456 to a subshell, it requires the System privilege, as well. This was done to
457 enhance manager security.
458 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
459 * New command: Atxfer. See doc/manager_1_1.txt for more details or
460 manager show command Atxfer from the CLI
461 * New command: IAXregistry. See doc/manager_1_1.txt for more details or
462 manager show command IAXregistry from the CLI
466 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
467 state in the dialplan, as well as creating custom device states that are
468 controllable from the dialplan.
469 * Extend CALLERID() function with "pres" and "ton" parameters to
470 fetch string representation of calling number presentation indicator
471 and numeric representation of type of calling number value.
472 * MailboxExists converted to dialplan function
473 * A new option to Dial() for telling IP phones not to count the call
474 as "missed" when dial times out and cancels.
475 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
476 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
477 held for any given channel. Also, locks are automatically freed when a
479 * Added HINT() dialplan function that allows retrieving hint information.
480 Hints are mappings between extensions and devices for the sake of
481 determining the state of an extension. This function can retrieve the list
482 of devices or the name associated with a hint.
483 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
485 * Added SYSINFO() dialplan function which allows retrieval of system information
486 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
487 the existence of a dialplan target.
488 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
489 upper and lower case, respectively.
490 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
491 ID for the call (not the Asterisk call ID or unique ID), provided that the
492 channel driver supports this. For SIP, you get the SIP call-ID for the
493 bridged channel which you can store in the CDR with a custom field.
497 * Added CLI permissions, config file: cli_permissions.conf
498 default is to allow all commands for every local user/group.
499 Also this new feature added three new CLI commands:
500 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
501 - cli reload permissions
502 - cli show permissions
503 * New CLI command "core show hint" (usage: core show hint <exten>)
504 * New CLI command "core show settings"
505 * Added 'core show channels count' CLI command.
506 * Added the ability to set the core debug and verbose values on a per-file basis.
507 * Added 'queue pause member' and 'queue unpause member' CLI commands
508 * Ability to set process limits ("ulimit") without restarting Asterisk
509 * Enhanced "agi debug" to print the channel name as a prefix to the debug
510 output to make debugging on busy systems much easier.
511 * New CLI commands "dialplan set extenpatternmatching true/false"
512 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
513 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
514 listed in the startup_commands section of cli.conf will get executed.
515 * Added a CLI command, "devstate change", which allows you to set custom device
516 states from the func_devstate module that provides the DEVICE_STATE() function
517 and handling of the "Custom:" devices.
518 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
519 sorted into the different possible callbacks, with the number of entries
520 currently scheduled for each. Gives you a feel for how busy the sip channel
522 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
523 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
524 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
528 * Improved NAT and STUN support.
529 chan_sip now can use port numbers in bindaddr, externip and externhost
530 options, as well as contact a STUN server to detect its external address
531 for the SIP socket. See sip.conf.sample, 'NAT' section.
532 * The default SIP useragent= identifier now includes the Asterisk version
533 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
534 If set, and the incoming request carries authentication info,
535 the username to match in the users list is taken from the Digest header
536 rather than from the From: field. This feature is considered experimental.
537 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
538 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
539 * The "localmask" setting was removed in version 1.2 and the reminder about it
540 being removed is now also removed.
541 * A new option "busylevel" for setting a level of calls where asterisk reports
542 a device as busy, to separate it from call-limit. This value is also added
543 to the SIP_PEER dialplan function.
544 * A new realtime family called "sipregs" is now supported to store SIP registration
545 data. If this family is defined, "sippeers" will be used for configuration and
546 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
547 registration data, as before.
548 * The SIPPEER function have new options for port address, call and pickup groups
549 * Added support for T.140 realtime text in SIP/RTP
550 * The "checkmwi" option has been removed from sip.conf, as it is no longer
551 required due to the restructuring of how MWI is handled. See the descriptions
552 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
553 for more information.
554 * Added rtpdest option to CHANNEL() dialplan function.
555 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
556 * SIP now adds a header to the CANCEL if the call was answered by another phone
557 in the same dial command, or if the new c option in dial() is used.
558 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
559 states it is not needed. For phones, however, that do require it the "registertrying" option
560 has been added so it can be enabled.
561 * A new option called "callcounter" (global/peer/user level) enables call counters needed
562 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
563 used to enable this functionality).
564 * New settings for timer T1 and timer B on a global level or per device. This makes it
565 possible to force timeout faster on non-responsive SIP servers. These settings are
566 considered advanced, so don't use them unless you have a problem.
567 * Added a dial string option to be able to set the To: header in an INVITE to any
569 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
570 the qualify frequency.
571 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
572 were not properly torn down due to network or endpoint failures during an established
574 * Added experimental TCP and TLS support for SIP. See doc/siptls.txt and
575 configs/sip.conf.sample for more information on how it is used.
576 * Added a new configuration option "authfailureevents" that enables manager events when
577 a peer can't authenticate properly.
578 * Added DNS manager support to registrations for peers not referencing a peer entry.
582 * Added the trunkmaxsize configuration option to chan_iax2.
583 * Added the srvlookup option to iax.conf
584 * Added support for OSP. The token is set and retrieved through the CHANNEL()
587 XMPP Google Talk/Jingle changes
588 -------------------------------
589 * Added the bindaddr option to gtalk.conf.
593 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
594 * Proper codec support in chan_skinny.
595 * Added settings for IP and Ethernet QoS requests
599 * Added separate settings for media QoS in mgcp.conf
601 Console Channel Driver changes
602 ------------------------------
603 * Added experimental support for video send & receive to chan_oss.
604 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
607 Phone channel changes (chan_phone)
608 ----------------------------------
609 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
611 H.323 channel Changes
612 ---------------------
613 * H323 remote hold notification support added (by NOTIFY message
614 and/or H.450 supplementary service)
616 Local channel changes
617 ---------------------
618 * The device state functionality in the Local channel driver has been updated
619 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
620 to just UNKNOWN if the extension exists.
621 * Added jitterbuffer support for chan_local. This allows you to use the
622 generic jitterbuffer on incoming calls going to Asterisk applications.
623 For example, this would allow you to use a jitterbuffer for an incoming
624 SIP call to Voicemail by putting a Local channel in the middle. This
625 feature is enabled by using the 'j' option in the Dial string to the Local
626 channel in conjunction with the existing 'n' option for local channels.
627 * A 'b' option has been added which causes chan_local to return the actual channel
628 that is behind it when queried. This is useful for transfer scenarios as the
629 actual channel will be transferred, not the Local channel.
631 Agent channel changes
632 ----------------------
633 * The ackcall and endcall options are now supplemented with options acceptdtmf
634 and enddtmf. These allow for the DTMF keypress to be configurable. The options
635 default to their old hard-coded values ('#' and '*' respectively) so this should
636 not break any existing agent installations.
638 DAHDI channel driver (chan_dahdi) Changes
639 ----------------------------------------
640 * SS7 support (via libss7 library)
641 * In India, some carriers transmit CID via dtmf. Some code has been added
642 that will handle some situations. The cidstart=polarity_IN choice has been added for
643 those carriers that transmit CID via dtmf after a polarity change.
644 * CID matching information is now shown when doing 'dialplan show'.
645 * Added dahdi show version CLI command.
646 * Added setvar support to chan_dahdi.conf channel entries.
647 * Added two new options: mwimonitor and mwimonitornotify. These options allow
648 you to enable MWI monitoring on FXO lines. When the MWI state changes,
649 the script specified in the mwimonitornotify option is executed. An internal
650 event indicating the new state of the mailbox is also generated, so that
651 the normal MWI facilities in Asterisk work as usual.
652 * Added signalling type 'auto', which attempts to use the same signalling type
653 for a channel as configured in DAHDI. This is primarily designed for analog
654 ports, but will also work for digital ports that are configured for FXS or FXO
655 signalling types. This mode is also the default now, so if your chan_dahdi.conf
656 does not specify signalling for a channel (which is unlikely as the sample
657 configuration file has always recommended specifying it for every channel) then
658 the 'auto' mode will be used for that channel if possible.
659 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
660 state for a channel; also ensured that the DNDState Manager event is
661 emitted no matter how the DND state is set or cleared.
665 * Added a new channel driver, chan_unistim. See doc/unistim.txt and
666 configs/unistim.conf.sample for details. This new channel driver allows
667 you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
668 * Added a new channel driver, chan_console, which uses portaudio as a cross
669 platform audio interface. It was written as a channel driver that would
670 work with Mac CoreAudio, but portaudio supports a number of other audio
671 interfaces, as well. Note that this channel driver requires v19 or higher
672 of portaudio; older versions have a different API.
676 * Added the ability to specify arguments to the Dial application when using
677 the DUNDi switch in the dialplan.
678 * Added the ability to set weights for responses dynamically. This can be
679 done using a global variable or a dialplan function. Using the SHELL()
680 function would allow you to have an external script set the weight for
682 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
683 functions will allow you to initiate a DUNDi query from the dialplan,
684 find out how many results there are, and access each one.
688 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
689 functions will allow you to initiate an ENUM lookup from the dialplan,
690 and Asterisk will cache the results. ENUMRESULT can be used to access
691 the results without doing multiple DNS queries.
695 * Added the ability to customize which sound files are used for some of the
696 prompts within the Voicemail application by changing them in voicemail.conf
697 * Added the ability for the "voicemail show users" CLI command to show users
698 configured by the dynamic realtime configuration method.
699 * MWI (Message Waiting Indication) handling has been significantly
700 restructured internally to Asterisk. It is now totally event based
701 instead of polling based. The voicemail application will notify other
702 modules that have subscribed to MWI events when something in the mailbox
704 This also means that if any other entity outside of Asterisk is changing
705 the contents of mailboxes, then the voicemail application still needs to
706 poll for changes. Examples of situations that would require this option
707 are web interfaces to voicemail or an email client in the case of using
708 IMAP storage. So, two new options have been added to voicemail.conf
709 to account for this: "pollmailboxes" and "pollfreq". See the sample
710 configuration file for details.
711 * Added "tw" language support
712 * Added support for storage of greetings using an IMAP server
713 * Added ability to customize forward, reverse, stop, and pause keys for message playback
714 * SMDI is now enabled in voicemail using the smdienable option.
715 * A "lockmode" option has been added to asterisk.conf to configure the file
716 locking method used for voicemail, and potentially other things in the
717 future. The default is the old behavior, lockfile. However, there is a
718 new method, "flock", that uses a different method for situations where the
719 lockfile will not work, such as on SMB/CIFS mounts.
720 * Added the ability to backup deleted messages, to ease recovery in the case
721 that a user accidentally deletes a message, and discovers that they need it.
722 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
723 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
724 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
725 voicemail boxes. The SMDI interface can also poll for MWI changes when some
726 outside entity is modifying the state of the mailbox (such as IMAP storage or
727 a web interface of some kind).
728 * Added the support for marking messages as "urgent." There are two methods to accomplish
729 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
730 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
731 the message as urgent after he has recorded a voicemail by following the voice instructions.
732 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
737 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
738 used across multiple queues.
739 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
740 setqueueentryvar options for each queue, see queues.conf.sample for details.
741 * Added keepstats option to queues.conf which will keep queue
742 statistics during a reload.
743 * setinterfacevar option in queues.conf also now sets a variable
744 called MEMBERNAME which contains the member's name.
745 * Added 'Strategy' field to manager event QueueParams which represents
746 the queue strategy in use.
747 * Added option to run macro when a queue member is connected to a caller,
748 see queues.conf.sample for details.
749 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
750 does not count paused queue members as unavailable.
751 * Added min-announce-frequency option to queues.conf which allows you to control the
752 minimum amount of time between queue announcements for use when the caller's queue
753 position changes frequently.
754 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
756 * Added ability for non-realtime queues to have realtime members
757 * Added the "linear" strategy to queues.
758 * Added the "wrandom" strategy to queues.
759 * Added new channel variable QUEUE_MIN_PENALTY
760 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
761 rules in queuerules.conf. See configs/queuerules.conf.sample for details
762 * Added a new parameter for member definition, called state_interface. This may be
763 used so that a member may be called via one interface but have a different interface's
764 device state reported.
765 * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
766 "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
767 "manager show command QueueReset."
768 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
769 specified by the periodic-announce option, then one will be chosen randomly when it is time
770 to play a periodic announcment
771 * New configuration options: announce-position now takes two more values in addition to "yes" and
772 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
773 announce-position-limit. By setting announce-position to "limit" callers will only have their
774 position announced if their position is less than what is specified by announce-position-limit.
775 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
776 will be told that their are more than announce-position-limit callers waiting.
777 * Two new queue log events have been added. An ADDMEMBER event will be logged
778 when a realtime queue member is added and a REMOVEMEMBER event will be logged
779 when a realtime queue member is removed. Since there is no calling channel associated
780 with these events, the string "REALTIME" is placed where the channel's unique id
782 * The configuration method for the "joinempty" and "leavewhenempty" options has
783 changed to a comma-separated list of methods of determining member availability
784 instead of vague terms such as "yes," "loose," "no," and "strict." These old four
785 values are still accepted for backwards-compatibility, though.
786 * The average talktime is now calculated on queues. This information is reported via the
787 CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
788 and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
793 * The 'o' option to provide an optimization has been removed and its functionality
794 has been enabled by default.
795 * When a conference is created, the UNIQUEID of the channel that caused it to be
796 created is stored. Then, every channel that joins the conference will have the
797 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
798 callers that come and go from long standing conferences.
799 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
800 except it does operations on a channel by name, instead of number in a conference.
801 This is a very useful feature in combination with the 'X' option to ChanSpy.
802 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
804 * Added new RealTime functionality to provide support for scheduled conferencing.
805 This includes optional messages to the caller if they attempt to join before
806 the schedule start time, or to allow the caller to join the conference early.
807 Also included is optional support for limiting the number of callers per
809 * Added the S() and L() options to the MeetMe application. These are pretty
810 much identical to the S() and L() options to Dial(). They let you set
811 timeouts for the conference, as well as have warning sounds played to
812 let the caller know how much time is left, and when it is running out.
813 * Added the ability to do "meetme concise" with the "meetme" CLI command.
814 This extends the concise capabilities of this CLI command to include
815 listing all conferences, instead of an addition to the other sub commands
816 for the "meetme" command.
817 * Added the ability to specify the music on hold class used to play into the
818 conference when there is only one member and the M option is used.
819 * Added MEETME_INFO dialplan function which provides a way to query
820 various properties of a Meetme conference.
822 Other Dialplan Application Changes
823 ----------------------------------
824 * Argument support for Gosub application
825 * From the to-do lists: straighten out the app timeout args:
826 Wait() app now really does 0.3 seconds- was truncating arg to an int.
827 WaitExten() same as Wait().
828 Congestion() - Now takes floating pt. argument.
829 Busy() - now takes floating pt. argument.
830 Read() - timeout now can be floating pt.
831 WaitForRing() now takes floating pt timeout arg.
832 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
833 * Added 's' option to Page application.
834 * Added an optional timeout argument to the Page application.
835 * Added 'E', 'V', and 'P' commands to ExternalIVR.
836 * Added 'o' and 'X' options to Chanspy.
837 * Added a new dialplan application, Bridge, which allows you to bridge the
838 calling channel to any other active channel on the system.
839 * Added the ability to specify a music on hold class to play instead of ringing
840 for the SLATrunk application.
841 * The Read application no longer exits the dialplan on error. Instead, it sets
842 READSTATUS to ERROR, which you can catch and handle separately.
843 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
844 of asking for verification of each name, one at a time.
845 * Privacy() no longer uses privacy.conf, as all options are specifyable as
846 direct options to the app.
847 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
849 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
850 * The ChannelRedirect application no longer exits the dialplan if the given channel
851 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
852 or NOCHANNEL if the given channel was not found.
853 * The silencethreshold setting that was previously configurable in multiple
854 applications is now settable globally via dsp.conf.
856 Music On Hold Changes
857 ---------------------
858 * A new option, "digit", has been added for music on hold classes in
859 musiconhold.conf. If this is set for a music on hold class, a caller
860 listening to music on hold can press this digit to switch to listening
861 to this music on hold class.
862 * Support for realtime music on hold has been added.
863 * In conjunction with the realtime music on hold, a general section has
864 been added to musiconhold.conf, its sole variable is cachertclasses. If this
865 is set, then music on hold classes found in realtime will be cached in memory.
869 * AEL upgraded to use the Gosub with Arguments instead
870 of Macro application, to hopefully reduce the problems
871 seen with the artificially low stack ceiling that
872 Macro bumps into. Macros can only call other Macros
873 to a depth of 7. Tests run using gosub, show depths
874 limited only by virtual memory. A small test demonstrated
875 recursive call depths of 100,000 without problems.
876 -- in addition to this, all apps that allowed a macro
877 to be called, as in Dial, queues, etc, are now allowing
878 a gosub call in similar fashion.
879 * AEL now generates LOCAL(argname) declarations when it
880 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
881 etc. That makes the arguments local in scope. The user
882 can define their own local variables in macros, now,
883 by saying "local myvar=someval;" or using Set() in this
884 fashion: Set(LOCAL(myvar)=someval); ("local" is now
886 * utils/conf2ael introduced. Will convert an extensions.conf
887 file into extensions.ael. Very crude and unfinished, but
888 will be improved as time goes by. Should be useful for a
889 first pass at conversion.
890 * aelparse will now read extensions.conf to see if a referenced
891 macro or context is there before issueing a warning.
892 * AEL parser sets a local channel variable ~~EXTEN~~, to
893 preserve the value of ${EXTEN} thru switch statements.
894 * New operator in $[...] expressions: the ~~ operator serves
895 as a concatenation operator. AT THE MOMENT, it is really only
896 necessary and useful in AEL, especially in if() expressions.
897 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
898 any enclosing double-quotes, and evaluate to the value of a
899 concatenated with the value of b. For example if a is set to
900 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
904 Call Features (res_features) Changes
905 ------------------------------------
906 * Added the parkedcalltransfers option to features.conf
907 * Added parkedcallparking option to control one touch parking w/ parking
909 * Added parkedcallhangup option to control disconnect feature w/ parking
911 * Added parkedcallrecording option to control one-touch record w/ parking
913 * Added BRIDGE_FEATURES variable to set available features for a channel
914 * The built-in method for doing attended transfers has been updated to
915 include some new options that allow you to have the transferee sent
916 back to the person that did the transfer if the transfer is not successful.
917 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
918 in features.conf.sample.
919 * Added support for configuring named groups of custom call features in
920 features.conf. This means that features can be written a single time, and
921 then mapped into groups of features for different key mappings or easier
923 * Updated the ParkedCall application to allow you to not specify a parking
924 extension. If you don't specify a parking space to pick up, it will grab
925 the first one available.
926 * Added cli command 'features reload' to reload call features from features.conf
927 * Moved into core asterisk binary.
928 * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
930 Language Support Changes
931 ------------------------
932 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
933 * Added support for the Hungarian language for saying numbers, dates, and times.
937 * Added SPEECH commands for speech recognition. A complete listing can be found
939 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
940 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
941 does not behave as expected; the native command needs to be used, instead.
945 * Added rotatestrategy option to logger.conf, along with two new options:
946 "timestamp" which will use the time to name the logger files instead of
947 sequence number; and "rotate", which rotates the names of the logfiles,
948 similar to the way syslog rotates files.
949 * Added exec_after_rotate option to logger.conf, which allows a system
950 command to be run after rotation. This is primarily useful with
951 rotatestrategry=rotate, to allow a limit on the number of logfiles kept
952 and to ensure that the oldest log file gets deleted.
953 * Added realtime support for the queue log
957 * The cdr_manager module has a [mappings] feature, like cdr_custom,
958 to add fields to the manager event from the CDR variables.
959 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
960 backend database CDR table. Specifically, additional, non-standard
961 columns are supported, merely by setting the corresponding CDR variable in
962 your dialplan. In addition, you may alias any column to another name (for
963 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
964 simply "alias src => ANI" in the configuration file). Records may be
965 posted to more than one backend, simply by specifying multiple categories
966 in the configuration file. And finally, you may filter which CDRs get
967 posted to each backend, by specifying a filter (which the record must
968 match) for the particular category. Filters are additive (meaning all
969 rules must match to post that CDR).
970 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
971 module. Specifically, you may add additional columns into the table and
972 they will be set, if you set the corresponding CDR variable name. Also,
973 if you omit columns in your database table, they will be silently skipped
974 (but a record will still be inserted, based on what columns remain). Note
975 that the other two features from cdr_adaptive_odbc (alias and filter) are
976 not currently supported.
977 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
978 has been disabled using the NoCDR application.
980 Miscellaneous New Modules
981 -------------------------
982 * Added a new CDR module, cdr_sqlite3_custom.
983 * Added a new realtime configuration module, res_config_sqlite
984 * Added a new codec translation module, codec_resample, which re-samples
985 signed linear audio between 8 kHz and 16 kHz to help support wideband
987 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
988 based on configuration templates that use Asterisk dialplan function and
989 variable substitution. It should be possible to create phone profiles and
990 templates that work for the majority of phones provisioned over http. It
991 is currently only intended to provision a single user account per phone.
992 An example profile and set of templates for Polycom phones is provided.
993 NOTE: Polycom firmware is not included, but should be placed in
994 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
995 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
996 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
997 provided; there is a JACK() application, and a JACK_HOOK() function. Both
998 interfaces create an input and output JACK port. The application makes
999 these ports the endpoint of the call. The audio coming from the channel
1000 goes out the output port and whatever comes back in on the input port is
1001 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
1002 audiohook on the channel. This lets you run the audio coming from a
1003 channel through JACK, and whatever comes back in is what gets forwarded
1004 on as the channel's audio. This is very useful for building custom
1005 vocoders or doing recording or analysis of the channel's audio in another
1007 * Added a new module, res_config_curl, which permits using a HTTP POST url
1008 to retrieve, create, update, and delete realtime information from a remote
1009 web server. Note that this module requires func_curl.so to be loaded for
1010 backend functionality.
1011 * Added a new module, res_config_ldap, which permits the use of an LDAP
1012 server for realtime data access.
1013 * Added support for writing and running your dialplan in lua using the pbx_lua
1014 module. See configs/extensions.lua.sample for examples of how to do this.
1018 * Ability to use libcap to set high ToS bits when non-root
1019 on Linux. If configure is unable to find libcap then you
1020 can use --with-cap to specify the path.
1021 * Added maxfiles option to options section of asterisk.conf which allows you to specify
1022 what Asterisk should set as the maximum number of open files when it loads.
1023 * Added the jittertargetextra configuration option.
1024 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
1025 configuration files for the IP channel drivers. The new option is "cos".
1026 This information is also documented in doc/qos.tex, or the IP Quality of Service
1027 section of asterisk.pdf.
1028 * When originating a call using AMI or pbx_spool that fails the reason for failure
1029 will now be available in the failed extension using the REASON dialplan variable.
1030 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
1031 It allows you to configure a prefix for auto-monitor recordings.
1032 * A new extension pattern matching algorithm, based on a trie, is introduced
1033 here, that could noticeably speed up mid-sized to large dialplans.
1034 It is NOT used by default, as duplicating the behaviour of the old pattern
1035 matcher is still under development. A config file option, in extensions.conf,
1036 in the [general] section, called "extenpatternmatchingnew", is by default
1037 set to false; setting that to true will force the use of the new algorithm.
1038 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
1039 be used to switch the algorithms at run time.
1040 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
1041 specifying which socket to use to connect to the running Asterisk daemon
1043 * Performance enhancements to the sched facility, which is used in
1044 the channel drivers, etc. Added hashtabs and doubly-linked lists
1045 to speed up deletion; start at the beginning or end of list to
1047 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
1048 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
1049 Added regression tests to the tests/ dir, also.
1050 * Added a refcount trace feature to astobj2 for those trying to balance
1051 object creation, deletion; work, play; space and time. See the
1052 notes in astobj2.h. Also, see utils/refcounter as well, as a
1053 quick way to find unbalanced refcounts in what could be a sea
1054 of objects that were balanced.
1055 * Added logging to 'make update' command. See update.log
1056 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
1057 do not come from the remote party.
1058 * Added the 'n' option to the SpeechBackground application to tell it to not
1059 answer the channel if it has not already been answered.
1060 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
1061 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
1063 * iLBC source code no longer included (see UPGRADE.txt for details)
1064 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
1065 deadlock is detected, a backtrace of the stack which led to the lock calls
1066 will be output to the CLI.
1067 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
1068 the "core show locks" CLI command will give lock information output as well
1069 as a backtrace of the stack which led to the lock calls.
1070 * users.conf now sports an optional alternateexts property, which permits
1071 allocation of additional extensions which will reach the specified user.
1072 * A new option for the configure script, --enable-internal-poll, has been added
1073 for use with systems which may have a buggy implementation of the poll system
1074 call. If you notice odd behavior such as the CLI being unresponsive on remote
1075 consoles, you may want to try using this option. This option is enabled by default
1076 on Darwin systems since it is known that the Darwin poll() implementation has
1080 --------------------
1081 * In addition to timing from DAHDI, there is a new timing module called
1082 res_timing_timerfd. In order to use this, you must be running Linux with
1083 a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
1084 script will be able to tell if you have the requirements. From menuselect, select
1085 res_timing_timerfd from the Resource Modules menu.
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