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 a function to remove SIP headers added in the dialplan before the
48 first INVITE is generated - SIPRemoveHeader()
49 * Channel variables set with setvar= in a device configuration is now
50 set both for inbound and outbound calls.
51 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
55 * The configuration file now holds separate sections for devices and lines.
56 Please have a look at configs/skinny.conf.sample and change your skinny.conf
61 * The UK option waitfordialtone has been added for use with BT analog
63 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
64 is used in conjunction with the 'faxdetect' configuration option. When
65 'faxbuffers' is used and fax tones are detected, the channel will dynamically
66 switch to the configured faxbuffers policy. For example, to use 6 buffers
67 and a 'full' buffer policy for a fax transmission, add:
69 The faxbuffers configuration will be in affect until the call is torn down.
73 * Added a new dialplan function, CURLOPT, which permits setting various
74 options that may be useful with the CURL dialplan function, such as
75 cookies, proxies, connection timeouts, passwords, etc.
76 * Permit the syntax and synopsis fields of the corresponding dialplan
77 functions to be individually set from func_odbc.conf.
78 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
79 * func_odbc now may specify an insert query to execute, when the write query
80 affects 0 rows (usually indicating that no such row exists).
81 * Added a new dialplan function, LISTFILTER, which permits removing elements
82 from a set list, by name. Uses the same general syntax as the existing CUT
83 and FIELDQTY dialplan functions, which also manage lists.
84 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
85 obtaining realtime data from the dialplan.
86 * Added LOCAL_PEEK, which I have no idea how to use, but Leif Madsen wanted it.
87 Russell says it's, like, a scope resolution function for LOCAL variables.
88 Totally. Hopefully, that means more to you than it does to me.
89 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
90 of "core show function AUDIOHOOK_INHERIT" from the CLI
91 * Added AES_ENCRYPT. For information on its use, please see the output
92 of "core show function AES_ENCRYPT" from the CLI
93 * Added AES_DECRYPT. For information on its use, please see the output
94 of "core show function AES_DECRYPT" from the CLI
98 * DAHDISendCallreroutingFacility parameters are now comma-separated,
99 instead of the old pipe.
100 * Scheduled meetme conferences may now have their end times extended by
102 * app_authenticate now gives the ability to select a prompt other than
104 * app_directory now pays attention to the searchcontexts setting in
105 voicemail.conf and will look through all contexts, if no context is
106 specified in the initial argument.
107 * A new application, Originate, has been introduced, that allows asynchronous
108 call origination from the dialplan.
112 * The Asterisk CLI has a new command, "channel redirect", which is similar in
113 operation to the AMI Redirect action.
114 * res_jabber: autoprune has been disabled by default, to avoid misconfiguration
115 that would end up being interpreted as a bug once Asterisk started removing
116 the contacts from a user list.
117 * extensions.conf now allows you to use keyword "same" to define an extension
118 without actually specifying an extension. It uses exactly the same pattern
119 as previously used on the last "exten" line. For example:
120 exten => 123,1,NoOp(something)
121 same => n,SomethingElse()
122 * musiconhold.conf classes of type 'files' can now use relative directory paths,
123 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
124 * All deprecated CLI commands are removed from the sourcecode. They are now handled
125 by the new clialiases module. See cli_aliases.conf.sample file.
126 * Times within timespecs are now accurate down to the minute. This is a change
127 from historical Asterisk, which only provided timespecs rounded to the nearest
128 even (read: evenly divisible by 2) minute mark.
129 * The realtime switch now supports an option flag, 'p', which disables searches for
131 * In addition to a time range and date range, timespecs now accept a 5th optional
132 argument, timezone. This allows you to perform time checks on alternate
133 timezones, especially if those daylight savings time ranges vary from your
134 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
136 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
137 give you the correct output for an asterisk box behind nat. It will give you the
138 externhost and localnet settings.
139 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
140 can connect calls in passthrough mode, as well as record and play back files.
142 Asterisk Manager Interface
143 --------------------------
144 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
145 a non-empty value) in your request. If you do this, any pending AMI events will
146 *not* be included in the response to your request as they would normally, but
147 will be left in the event queue for the next request you make to retrieve. For
148 some applications, this will allow you to guarantee that you will only see
149 events in responses to 'WaitEvent' actions, and can better know when to expect them.
150 To know whether the Asterisk server supports this header or not, your client can
151 inspect the first response back from the server to see if it includes this header:
153 Pragma: SuppressEvents
155 If this is included, the server supports event suppression.
157 ------------------------------------------------------------------------------
158 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
159 ------------------------------------------------------------------------------
161 Device State Handling
162 ---------------------
163 * The event infrastructure in Asterisk got another big update to help support
164 distributed events. It currently supports distributed device state and
165 distributed Voicemail MWI (Message Waiting Indication). A new module has
166 been merged, res_ais, which facilitates communicating events between servers.
167 It uses the SAForum AIS (Service Availability Forum Application Interface
168 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
169 a cluster of Asterisk servers, and to share events between them. For more
170 information on setting this up, see doc/distributed_devstate.txt.
174 * Added a new dialplan function, AST_CONFIG(), which allows you to access
175 variables from an Asterisk configuration file.
176 * The JACK_HOOK function now has a c() option to supply a custom client name.
177 * Added two new dialplan functions from libspeex for audio gain control and
178 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
179 rx directions of a channel from the dialplan.
180 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
181 based on other parameters. The default is still to search based on the
182 forwarding station ID. However, there are new options that allow you to search
183 based on the message desk terminal ID, or the message desk number.
184 * TIMEOUT() has been modified to be accurate down to the millisecond.
185 * ENUM*() functions now include the following new options:
186 - 'u' returns the full URI and does not strip off the URI-scheme.
187 - 's' triggers ISN specific rewriting
188 - 'i' looks for branches into an Infrastructure ENUM tree
189 - 'd' for a direct DNS lookup without any flipping of digits.
190 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
191 * CHANNEL() now has options for the maximum, minimum, and standard or normal
192 deviation of jitter, rtt, and loss for a call using chan_sip.
194 DAHDI channel driver (chan_dahdi) Changes
195 ----------------------------------------
196 * Channels can now be configured using named sections in chan_dahdi.conf, just
197 like other channel drivers, including the use of templates.
198 * The default for pridialplan has changed from 'national' to 'unknown'.
202 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
203 to something that matches the pattern a hint will be created using the contents
204 and variables evaluated.
205 * Dialplan matching has been extended to allow an extension to return to the
206 PBX core to wait for more digits. This is done by using the new dialplan
207 application called "Incomplete". This will permit a whole new level of
208 extension control, by giving the administrator more control over early
209 matches employing one of the short-circuit pattern match operators. Note
210 that custom applications can trigger this same behavior by returning the
211 special value AST_PBX_INCOMPLETE.
215 * Directory now permits both first and last names to be matched at the same
216 time. In addition, the number of digits to enter of the name can be set in
217 the arguments to Directory; previously, you could enter only 3, regardless
218 of how many names are in your company. For large companies, this should be
220 * Voicemail now permits a mailbox setting to wrap around from first to last
221 messages, if the "messagewrap" option is set to a true value.
222 * Voicemail now permits an external script to be run, for password validation.
223 The script should output "VALID" or "INVALID" on stdout, depending upon the
224 wish to validate or invalidate the password given. Arguments are:
225 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
227 * Dial has a new option: F(context^extension^pri), which permits a callee to
228 continue in the dialplan, at the specified label, if the caller hangs up.
229 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
230 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
231 * The Jack application now has a c() option to supply a custom client name.
232 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
233 like the pre-existing whisper mode, except that the spy can also talk to the
234 participant on the bridged channel as well.
235 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
236 to be spoken instead of the channel name or number. For more information on the
237 use of this option, issue the command "core show application ChanSpy" from the
239 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
240 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
241 words, if using the 'd' option, it is not possible to enter a number to append to
242 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
243 change to whisper mode, and pressing 6 will change to barge mode.
244 * ExternalIVR now takes several options that affect the way it performs, as
245 well as having several new commands. Please see doc/externalivr.txt for the
246 complete documentation.
247 * Added ability to communicate over a TCP socket instead of forking a child process for the
248 ExternalIVR application.
249 * ChanIsAvail has a new option, 'a', which will return all available channels instead
250 of just the first one if you give the function more then one channel to check.
251 * PrivacyManager now takes an option where you can specify a context where the
252 given number will be matched. This way you have more control over who is allowed
253 and it stops the people who blindly enter 10 digits.
254 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
255 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
256 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
257 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
258 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
259 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
260 * The Dial() application no longer copies the language used by the caller to the callee's
261 channel. If you desire for the caller's channel's language to be used for file playback
262 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
263 * SendImage() no longer hangs up the channel on error; instead, it sets the
264 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
265 'UNSUPPORTED'. This change makes SendImage() more consistent with other
267 * Park has a new option, 's', which silences the announcement of the parking space number.
268 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
269 invalid input and will be assumed to mean that no timeout is desired.
273 * Added DNS manager support to registrations for peers referencing peer entries.
274 DNS manager runs in the background which allows DNS lookups to be run asynchronously
275 as well as periodically updating the IP address. These properties allow for
276 better performance as well as recovery in the event of an IP change.
277 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
278 load/reload of large numbers of peers/users by ~40x (for large lists of peers.
279 Initially, we saw 4x improvement in call setup/destruction, but at the time
280 of merging, this gain has disappeared; further research will be done to try
281 and restore this performance improvement. Astobj2 refcounting is now used
282 for users, peers, and dialogs. Users are encouraged to assist in regression
283 testing and problem reporting!
284 * Added ability to specify registration expiry time on a per registration basis in
286 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
288 * Added t38pt_usertpsource option. See sip.conf.sample for details.
289 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
290 * 'sip show peers' and 'sip show users' display their entries sorted in
291 alphabetical order, as opposed to the order they were in, in the config
293 * Videosupport now supports an additional option, "always", which always sets
294 up video RTP ports, even on clients that don't support it. This helps with
295 callfiles and certain transfers to ensure that if two video phones are
296 connected, they will always share video feeds.
300 * Existing DNS manager lookups extended to check for SRV records.
301 * IAX2 encryption support has been improved to support periodic key rotation
302 within a call for enhanced security. The option "keyrotate" has been
303 provided to disable this functionality to preserve backwards compatibility
304 with older versions of IAX2 that do not support key rotation.
308 * New CLI command, "config reload <file.conf>" which reloads any module that
309 references that particular configuration file. Also added "config list"
310 which shows which configuration files are in use.
311 * New CLI commands, "pri show version" and "ss7 show version" that will
312 display which version of libpri and libss7 are being used, respectively.
313 A new API call was added so trunk will now have to be compiled against
314 a versions of libpri and libss7 that have them or it will not know that
315 these libraries exist.
316 * The commands "core show globals", "core set global" and "core set chanvar" has
317 been deprecated in favor of the more semanticly correct "dialplan show globals",
318 "dialplan set chanvar" and "dialplan set global".
319 * New CLI command "dialplan show chanvar" to list all variables associated
320 with a given channel.
324 * Addresses managed by DNS manager now can check to see if there is a DNS
325 SRV record for a given domain and will use that hostname/port if present.
327 AMI - The manager (TCP/TLS/HTTP)
328 --------------------------------
329 * The Status command now takes an optional list of variables to display
330 along with channel status.
331 * The QueueEntry event now also includes the channel's uniqueid
335 * res_odbc no longer has a limit of 1023 total possible unshared connections,
336 as some people were running into this limit. This limit has been increased
341 * The TRANSFER queue log entry now includes the the caller's original
342 position in the transferred-from queue.
343 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
344 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
345 as well as an explanation about timeout options in general
346 * Added a new option - C - for forcing the "answered elsewhere" flag on
347 cancellation of calls in to members of the queue. This is to avoid the
348 call to a member of a queue having the call listed as a "missed call".
352 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
353 adaptive capabilities. What this means in practical terms is that if your
354 realtime table lacks critical fields, Asterisk will now emit warnings to
355 that effect. Also, some of the realtime drivers have the ability (if
356 configured) to automatically add those columns to the table with the
357 correct type and length.
361 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
362 the 'setvar' option to cause a given audio file to be played upon completion
363 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
364 Skinny channels only.
365 * You can now compile Asterisk against the Hoard Memory Allocator, see doc/hoard.txt
366 for more information.
367 * Config file variables may now be appended to, by using the '+=' append
368 operator. This is most helpful when working with long SQL queries in
369 func_odbc.conf, as the queries no longer need to be specified on a single
371 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
372 which will add a second to the billsec when the ending
373 time is set, if the number in the microseconds field of the end time is
374 greater than the number of microseconds in the answer time. This allows
375 users to count the 'initiated' seconds in their billing records.
377 ------------------------------------------------------------------------------
378 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
379 ------------------------------------------------------------------------------
381 AMI - The manager (TCP/TLS/HTTP)
382 --------------------------------
383 * Manager has undergone a lot of changes, all of them documented
384 in doc/manager_1_1.txt
385 * Manager version has changed to 1.1
386 * Added a new action 'CoreShowChannels' to list currently defined channels
387 and some information about them.
388 * Added a new action 'SIPshowregistry' to list SIP registrations.
389 * Added TLS support for the manager interface and HTTP server
390 * Added the URI redirect option for the built-in HTTP server
391 * The output of CallerID in Manager events is now more consistent.
392 CallerIDNum is used for number and CallerIDName for name.
393 * Enable https support for builtin web server.
394 See configs/http.conf.sample for details.
395 * Added a new action, GetConfigJSON, which can return the contents of an
396 Asterisk configuration file in JSON format. This is intended to help
397 improve the performance of AJAX applications using the manager interface
399 * SIP and IAX manager events now use "ChannelType" in all cases where we
400 indicate channel driver. Previously, we used a mixture of "Channel"
401 and "ChannelDriver" headers.
402 * Added a "Bridge" action which allows you to bridge any two channels that
403 are currently active on the system.
404 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
405 the voicemail users setup.
406 * Added 'DBDel' and 'DBDelTree' manager commands.
407 * cdr_manager now reports events via the "cdr" level, separating it from
408 the very verbose "call" level.
409 * Manager users are now stored in memory. If you change the manager account
410 list (delete or add accounts) you need to reload manager.
411 * Added Masquerade manager event for when a masquerade happens between
413 * Added "manager reload" command for the CLI
414 * Lots of commands that only provided information are now allowed under the
415 Reporting privilege, instead of only under Call or System.
416 * The IAX* commands now require either System or Reporting privilege, to
417 mirror the privileges of the SIP* commands.
418 * Added ability to retrieve list of categories in a config file.
419 * Added ability to retrieve the content of a particular category.
420 * Added ability to empty a context.
421 * Created new action to create a new file.
422 * Updated delete action to allow deletion by line number with respect to category.
423 * Added new action insert to add new variable to category at specified line.
424 * Updated action newcat to allow new category to be inserted in file above another
426 * Added new event "JitterBufStats" in the IAX2 channel
427 * Originate now requires the Originate privilege and, if you want to call out
428 to a subshell, it requires the System privilege, as well. This was done to
429 enhance manager security.
430 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
431 * New command: Atxfer. See doc/manager_1_1.txt for more details or
432 manager show command Atxfer from the CLI
433 * New command: IAXregistry. See doc/manager_1_1.txt for more details or
434 manager show command IAXregistry from the CLI
438 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
439 state in the dialplan, as well as creating custom device states that are
440 controllable from the dialplan.
441 * Extend CALLERID() function with "pres" and "ton" parameters to
442 fetch string representation of calling number presentation indicator
443 and numeric representation of type of calling number value.
444 * MailboxExists converted to dialplan function
445 * A new option to Dial() for telling IP phones not to count the call
446 as "missed" when dial times out and cancels.
447 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
448 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
449 held for any given channel. Also, locks are automatically freed when a
451 * Added HINT() dialplan function that allows retrieving hint information.
452 Hints are mappings between extensions and devices for the sake of
453 determining the state of an extension. This function can retrieve the list
454 of devices or the name associated with a hint.
455 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
457 * Added SYSINFO() dialplan function which allows retrieval of system information
458 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
459 the existence of a dialplan target.
460 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
461 upper and lower case, respectively.
462 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
463 ID for the call (not the Asterisk call ID or unique ID), provided that the
464 channel driver supports this. For SIP, you get the SIP call-ID for the
465 bridged channel which you can store in the CDR with a custom field.
469 * Added CLI permissions, config file: cli_permissions.conf
470 default is to allow all commands for every local user/group.
471 Also this new feature added three new CLI commands:
472 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
473 - cli reload permissions
474 - cli show permissions
475 * New CLI command "core show hint" (usage: core show hint <exten>)
476 * New CLI command "core show settings"
477 * Added 'core show channels count' CLI command.
478 * Added the ability to set the core debug and verbose values on a per-file basis.
479 * Added 'queue pause member' and 'queue unpause member' CLI commands
480 * Ability to set process limits ("ulimit") without restarting Asterisk
481 * Enhanced "agi debug" to print the channel name as a prefix to the debug
482 output to make debugging on busy systems much easier.
483 * New CLI commands "dialplan set extenpatternmatching true/false"
484 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
485 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
486 listed in the startup_commands section of cli.conf will get executed.
487 * Added a CLI command, "devstate change", which allows you to set custom device
488 states from the func_devstate module that provides the DEVICE_STATE() function
489 and handling of the "Custom:" devices.
490 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
491 sorted into the different possible callbacks, with the number of entries
492 currently scheduled for each. Gives you a feel for how busy the sip channel
494 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
495 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
496 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
500 * Improved NAT and STUN support.
501 chan_sip now can use port numbers in bindaddr, externip and externhost
502 options, as well as contact a STUN server to detect its external address
503 for the SIP socket. See sip.conf.sample, 'NAT' section.
504 * The default SIP useragent= identifier now includes the Asterisk version
505 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
506 If set, and the incoming request carries authentication info,
507 the username to match in the users list is taken from the Digest header
508 rather than from the From: field. This feature is considered experimental.
509 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
510 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
511 * The "localmask" setting was removed in version 1.2 and the reminder about it
512 being removed is now also removed.
513 * A new option "busylevel" for setting a level of calls where asterisk reports
514 a device as busy, to separate it from call-limit. This value is also added
515 to the SIP_PEER dialplan function.
516 * A new realtime family called "sipregs" is now supported to store SIP registration
517 data. If this family is defined, "sippeers" will be used for configuration and
518 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
519 registration data, as before.
520 * The SIPPEER function have new options for port address, call and pickup groups
521 * Added support for T.140 realtime text in SIP/RTP
522 * The "checkmwi" option has been removed from sip.conf, as it is no longer
523 required due to the restructuring of how MWI is handled. See the descriptions
524 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
525 for more information.
526 * Added rtpdest option to CHANNEL() dialplan function.
527 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
528 * SIP now adds a header to the CANCEL if the call was answered by another phone
529 in the same dial command, or if the new c option in dial() is used.
530 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
531 states it is not needed. For phones, however, that do require it the "registertrying" option
532 has been added so it can be enabled.
533 * A new option called "callcounter" (global/peer/user level) enables call counters needed
534 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
535 used to enable this functionality).
536 * New settings for timer T1 and timer B on a global level or per device. This makes it
537 possible to force timeout faster on non-responsive SIP servers. These settings are
538 considered advanced, so don't use them unless you have a problem.
539 * Added a dial string option to be able to set the To: header in an INVITE to any
541 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
542 the qualify frequency.
543 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
544 were not properly torn down due to network or endpoint failures during an established
546 * Added experimental TCP and TLS support for SIP. See doc/siptls.txt and
547 configs/sip.conf.sample for more information on how it is used.
548 * Added a new configuration option "authfailureevents" that enables manager events when
549 a peer can't authenticate properly.
550 * Added DNS manager support to registrations for peers not referencing a peer entry.
554 * Added the trunkmaxsize configuration option to chan_iax2.
555 * Added the srvlookup option to iax.conf
556 * Added support for OSP. The token is set and retrieved through the CHANNEL()
558 * Added immediate option to iax.conf
559 * Added forceencryption option to iax.conf
561 XMPP Google Talk/Jingle changes
562 -------------------------------
563 * Added the bindaddr option to gtalk.conf.
567 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
568 * Proper codec support in chan_skinny.
569 * Added settings for IP and Ethernet QoS requests
573 * Added separate settings for media QoS in mgcp.conf
575 Console Channel Driver changes
576 ------------------------------
577 * Added experimental support for video send & receive to chan_oss.
578 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
581 Phone channel changes (chan_phone)
582 ----------------------------------
583 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
585 H.323 channel Changes
586 ---------------------
587 * H323 remote hold notification support added (by NOTIFY message
588 and/or H.450 supplementary service)
590 Local channel changes
591 ---------------------
592 * The device state functionality in the Local channel driver has been updated
593 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
594 to just UNKNOWN if the extension exists.
595 * Added jitterbuffer support for chan_local. This allows you to use the
596 generic jitterbuffer on incoming calls going to Asterisk applications.
597 For example, this would allow you to use a jitterbuffer for an incoming
598 SIP call to Voicemail by putting a Local channel in the middle. This
599 feature is enabled by using the 'j' option in the Dial string to the Local
600 channel in conjunction with the existing 'n' option for local channels.
601 * A 'b' option has been added which causes chan_local to return the actual channel
602 that is behind it when queried. This is useful for transfer scenarios as the
603 actual channel will be transferred, not the Local channel.
605 Agent channel changes
606 ----------------------
607 * The ackcall and endcall options are now supplemented with options acceptdtmf
608 and enddtmf. These allow for the DTMF keypress to be configurable. The options
609 default to their old hard-coded values ('#' and '*' respectively) so this should
610 not break any existing agent installations.
612 DAHDI channel driver (chan_dahdi) Changes
613 ----------------------------------------
614 * SS7 support (via libss7 library)
615 * In India, some carriers transmit CID via dtmf. Some code has been added
616 that will handle some situations. The cidstart=polarity_IN choice has been added for
617 those carriers that transmit CID via dtmf after a polarity change.
618 * CID matching information is now shown when doing 'dialplan show'.
619 * Added dahdi show version CLI command.
620 * Added setvar support to chan_dahdi.conf channel entries.
621 * Added two new options: mwimonitor and mwimonitornotify. These options allow
622 you to enable MWI monitoring on FXO lines. When the MWI state changes,
623 the script specified in the mwimonitornotify option is executed. An internal
624 event indicating the new state of the mailbox is also generated, so that
625 the normal MWI facilities in Asterisk work as usual.
626 * Added signalling type 'auto', which attempts to use the same signalling type
627 for a channel as configured in DAHDI. This is primarily designed for analog
628 ports, but will also work for digital ports that are configured for FXS or FXO
629 signalling types. This mode is also the default now, so if your chan_dahdi.conf
630 does not specify signalling for a channel (which is unlikely as the sample
631 configuration file has always recommended specifying it for every channel) then
632 the 'auto' mode will be used for that channel if possible.
633 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
634 state for a channel; also ensured that the DNDState Manager event is
635 emitted no matter how the DND state is set or cleared.
639 * Added a new channel driver, chan_unistim. See doc/unistim.txt and
640 configs/unistim.conf.sample for details. This new channel driver allows
641 you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
642 * Added a new channel driver, chan_console, which uses portaudio as a cross
643 platform audio interface. It was written as a channel driver that would
644 work with Mac CoreAudio, but portaudio supports a number of other audio
645 interfaces, as well. Note that this channel driver requires v19 or higher
646 of portaudio; older versions have a different API.
650 * Added the ability to specify arguments to the Dial application when using
651 the DUNDi switch in the dialplan.
652 * Added the ability to set weights for responses dynamically. This can be
653 done using a global variable or a dialplan function. Using the SHELL()
654 function would allow you to have an external script set the weight for
656 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
657 functions will allow you to initiate a DUNDi query from the dialplan,
658 find out how many results there are, and access each one.
662 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
663 functions will allow you to initiate an ENUM lookup from the dialplan,
664 and Asterisk will cache the results. ENUMRESULT can be used to access
665 the results without doing multiple DNS queries.
669 * Added the ability to customize which sound files are used for some of the
670 prompts within the Voicemail application by changing them in voicemail.conf
671 * Added the ability for the "voicemail show users" CLI command to show users
672 configured by the dynamic realtime configuration method.
673 * MWI (Message Waiting Indication) handling has been significantly
674 restructured internally to Asterisk. It is now totally event based
675 instead of polling based. The voicemail application will notify other
676 modules that have subscribed to MWI events when something in the mailbox
678 This also means that if any other entity outside of Asterisk is changing
679 the contents of mailboxes, then the voicemail application still needs to
680 poll for changes. Examples of situations that would require this option
681 are web interfaces to voicemail or an email client in the case of using
682 IMAP storage. So, two new options have been added to voicemail.conf
683 to account for this: "pollmailboxes" and "pollfreq". See the sample
684 configuration file for details.
685 * Added "tw" language support
686 * Added support for storage of greetings using an IMAP server
687 * Added ability to customize forward, reverse, stop, and pause keys for message playback
688 * SMDI is now enabled in voicemail using the smdienable option.
689 * A "lockmode" option has been added to asterisk.conf to configure the file
690 locking method used for voicemail, and potentially other things in the
691 future. The default is the old behavior, lockfile. However, there is a
692 new method, "flock", that uses a different method for situations where the
693 lockfile will not work, such as on SMB/CIFS mounts.
694 * Added the ability to backup deleted messages, to ease recovery in the case
695 that a user accidentally deletes a message, and discovers that they need it.
696 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
697 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
698 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
699 voicemail boxes. The SMDI interface can also poll for MWI changes when some
700 outside entity is modifying the state of the mailbox (such as IMAP storage or
701 a web interface of some kind).
702 * Added the support for marking messages as "urgent." There are two methods to accomplish
703 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
704 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
705 the message as urgent after he has recorded a voicemail by following the voice instructions.
706 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
711 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
712 used across multiple queues.
713 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
714 setqueueentryvar options for each queue, see queues.conf.sample for details.
715 * Added keepstats option to queues.conf which will keep queue
716 statistics during a reload.
717 * setinterfacevar option in queues.conf also now sets a variable
718 called MEMBERNAME which contains the member's name.
719 * Added 'Strategy' field to manager event QueueParams which represents
720 the queue strategy in use.
721 * Added option to run macro when a queue member is connected to a caller,
722 see queues.conf.sample for details.
723 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
724 does not count paused queue members as unavailable.
725 * Added min-announce-frequency option to queues.conf which allows you to control the
726 minimum amount of time between queue announcements for use when the caller's queue
727 position changes frequently.
728 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
730 * Added ability for non-realtime queues to have realtime members
731 * Added the "linear" strategy to queues.
732 * Added the "wrandom" strategy to queues.
733 * Added new channel variable QUEUE_MIN_PENALTY
734 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
735 rules in queuerules.conf. See configs/queuerules.conf.sample for details
736 * Added a new parameter for member definition, called state_interface. This may be
737 used so that a member may be called via one interface but have a different interface's
738 device state reported.
739 * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
740 "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
741 "manager show command QueueReset."
742 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
743 specified by the periodic-announce option, then one will be chosen randomly when it is time
744 to play a periodic announcment
745 * New configuration options: announce-position now takes two more values in addition to "yes" and
746 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
747 announce-position-limit. By setting announce-position to "limit" callers will only have their
748 position announced if their position is less than what is specified by announce-position-limit.
749 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
750 will be told that their are more than announce-position-limit callers waiting.
751 * Two new queue log events have been added. An ADDMEMBER event will be logged
752 when a realtime queue member is added and a REMOVEMEMBER event will be logged
753 when a realtime queue member is removed. Since there is no calling channel associated
754 with these events, the string "REALTIME" is placed where the channel's unique id
756 * The configuration method for the "joinempty" and "leavewhenempty" options has
757 changed to a comma-separated list of methods of determining member availability
758 instead of vague terms such as "yes," "loose," "no," and "strict." These old four
759 values are still accepted for backwards-compatibility, though.
760 * The average talktime is now calculated on queues. This information is reported via the
761 CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
762 and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
767 * The 'o' option to provide an optimization has been removed and its functionality
768 has been enabled by default.
769 * When a conference is created, the UNIQUEID of the channel that caused it to be
770 created is stored. Then, every channel that joins the conference will have the
771 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
772 callers that come and go from long standing conferences.
773 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
774 except it does operations on a channel by name, instead of number in a conference.
775 This is a very useful feature in combination with the 'X' option to ChanSpy.
776 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
778 * Added new RealTime functionality to provide support for scheduled conferencing.
779 This includes optional messages to the caller if they attempt to join before
780 the schedule start time, or to allow the caller to join the conference early.
781 Also included is optional support for limiting the number of callers per
783 * Added the S() and L() options to the MeetMe application. These are pretty
784 much identical to the S() and L() options to Dial(). They let you set
785 timeouts for the conference, as well as have warning sounds played to
786 let the caller know how much time is left, and when it is running out.
787 * Added the ability to do "meetme concise" with the "meetme" CLI command.
788 This extends the concise capabilities of this CLI command to include
789 listing all conferences, instead of an addition to the other sub commands
790 for the "meetme" command.
791 * Added the ability to specify the music on hold class used to play into the
792 conference when there is only one member and the M option is used.
793 * Added MEETME_INFO dialplan function which provides a way to query
794 various properties of a Meetme conference.
796 Other Dialplan Application Changes
797 ----------------------------------
798 * Argument support for Gosub application
799 * From the to-do lists: straighten out the app timeout args:
800 Wait() app now really does 0.3 seconds- was truncating arg to an int.
801 WaitExten() same as Wait().
802 Congestion() - Now takes floating pt. argument.
803 Busy() - now takes floating pt. argument.
804 Read() - timeout now can be floating pt.
805 WaitForRing() now takes floating pt timeout arg.
806 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
807 * Added 's' option to Page application.
808 * Added an optional timeout argument to the Page application.
809 * Added 'E', 'V', and 'P' commands to ExternalIVR.
810 * Added 'o' and 'X' options to Chanspy.
811 * Added a new dialplan application, Bridge, which allows you to bridge the
812 calling channel to any other active channel on the system.
813 * Added the ability to specify a music on hold class to play instead of ringing
814 for the SLATrunk application.
815 * The Read application no longer exits the dialplan on error. Instead, it sets
816 READSTATUS to ERROR, which you can catch and handle separately.
817 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
818 of asking for verification of each name, one at a time.
819 * Privacy() no longer uses privacy.conf, as all options are specifyable as
820 direct options to the app.
821 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
823 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
824 * The ChannelRedirect application no longer exits the dialplan if the given channel
825 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
826 or NOCHANNEL if the given channel was not found.
827 * The silencethreshold setting that was previously configurable in multiple
828 applications is now settable globally via dsp.conf.
830 Music On Hold Changes
831 ---------------------
832 * A new option, "digit", has been added for music on hold classes in
833 musiconhold.conf. If this is set for a music on hold class, a caller
834 listening to music on hold can press this digit to switch to listening
835 to this music on hold class.
836 * Support for realtime music on hold has been added.
837 * In conjunction with the realtime music on hold, a general section has
838 been added to musiconhold.conf, its sole variable is cachertclasses. If this
839 is set, then music on hold classes found in realtime will be cached in memory.
843 * AEL upgraded to use the Gosub with Arguments instead
844 of Macro application, to hopefully reduce the problems
845 seen with the artificially low stack ceiling that
846 Macro bumps into. Macros can only call other Macros
847 to a depth of 7. Tests run using gosub, show depths
848 limited only by virtual memory. A small test demonstrated
849 recursive call depths of 100,000 without problems.
850 -- in addition to this, all apps that allowed a macro
851 to be called, as in Dial, queues, etc, are now allowing
852 a gosub call in similar fashion.
853 * AEL now generates LOCAL(argname) declarations when it
854 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
855 etc. That makes the arguments local in scope. The user
856 can define their own local variables in macros, now,
857 by saying "local myvar=someval;" or using Set() in this
858 fashion: Set(LOCAL(myvar)=someval); ("local" is now
860 * utils/conf2ael introduced. Will convert an extensions.conf
861 file into extensions.ael. Very crude and unfinished, but
862 will be improved as time goes by. Should be useful for a
863 first pass at conversion.
864 * aelparse will now read extensions.conf to see if a referenced
865 macro or context is there before issueing a warning.
866 * AEL parser sets a local channel variable ~~EXTEN~~, to
867 preserve the value of ${EXTEN} thru switch statements.
868 * New operator in $[...] expressions: the ~~ operator serves
869 as a concatenation operator. AT THE MOMENT, it is really only
870 necessary and useful in AEL, especially in if() expressions.
871 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
872 any enclosing double-quotes, and evaluate to the value of a
873 concatenated with the value of b. For example if a is set to
874 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
878 Call Features (res_features) Changes
879 ------------------------------------
880 * Added the parkedcalltransfers option to features.conf
881 * Added parkedcallparking option to control one touch parking w/ parking
883 * Added parkedcallhangup option to control disconnect feature w/ parking
885 * Added parkedcallrecording option to control one-touch record w/ parking
887 * Added BRIDGE_FEATURES variable to set available features for a channel
888 * The built-in method for doing attended transfers has been updated to
889 include some new options that allow you to have the transferee sent
890 back to the person that did the transfer if the transfer is not successful.
891 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
892 in features.conf.sample.
893 * Added support for configuring named groups of custom call features in
894 features.conf. This means that features can be written a single time, and
895 then mapped into groups of features for different key mappings or easier
897 * Updated the ParkedCall application to allow you to not specify a parking
898 extension. If you don't specify a parking space to pick up, it will grab
899 the first one available.
900 * Added cli command 'features reload' to reload call features from features.conf
901 * Moved into core asterisk binary.
902 * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
904 Language Support Changes
905 ------------------------
906 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
907 * Added support for the Hungarian language for saying numbers, dates, and times.
911 * Added SPEECH commands for speech recognition. A complete listing can be found
913 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
914 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
915 does not behave as expected; the native command needs to be used, instead.
919 * Added rotatestrategy option to logger.conf, along with two new options:
920 "timestamp" which will use the time to name the logger files instead of
921 sequence number; and "rotate", which rotates the names of the logfiles,
922 similar to the way syslog rotates files.
923 * Added exec_after_rotate option to logger.conf, which allows a system
924 command to be run after rotation. This is primarily useful with
925 rotatestrategry=rotate, to allow a limit on the number of logfiles kept
926 and to ensure that the oldest log file gets deleted.
927 * Added realtime support for the queue log
931 * The cdr_manager module has a [mappings] feature, like cdr_custom,
932 to add fields to the manager event from the CDR variables.
933 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
934 backend database CDR table. Specifically, additional, non-standard
935 columns are supported, merely by setting the corresponding CDR variable in
936 your dialplan. In addition, you may alias any column to another name (for
937 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
938 simply "alias src => ANI" in the configuration file). Records may be
939 posted to more than one backend, simply by specifying multiple categories
940 in the configuration file. And finally, you may filter which CDRs get
941 posted to each backend, by specifying a filter (which the record must
942 match) for the particular category. Filters are additive (meaning all
943 rules must match to post that CDR).
944 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
945 module. Specifically, you may add additional columns into the table and
946 they will be set, if you set the corresponding CDR variable name. Also,
947 if you omit columns in your database table, they will be silently skipped
948 (but a record will still be inserted, based on what columns remain). Note
949 that the other two features from cdr_adaptive_odbc (alias and filter) are
950 not currently supported.
951 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
952 has been disabled using the NoCDR application.
954 Miscellaneous New Modules
955 -------------------------
956 * Added a new CDR module, cdr_sqlite3_custom.
957 * Added a new realtime configuration module, res_config_sqlite
958 * Added a new codec translation module, codec_resample, which re-samples
959 signed linear audio between 8 kHz and 16 kHz to help support wideband
961 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
962 based on configuration templates that use Asterisk dialplan function and
963 variable substitution. It should be possible to create phone profiles and
964 templates that work for the majority of phones provisioned over http. It
965 is currently only intended to provision a single user account per phone.
966 An example profile and set of templates for Polycom phones is provided.
967 NOTE: Polycom firmware is not included, but should be placed in
968 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
969 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
970 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
971 provided; there is a JACK() application, and a JACK_HOOK() function. Both
972 interfaces create an input and output JACK port. The application makes
973 these ports the endpoint of the call. The audio coming from the channel
974 goes out the output port and whatever comes back in on the input port is
975 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
976 audiohook on the channel. This lets you run the audio coming from a
977 channel through JACK, and whatever comes back in is what gets forwarded
978 on as the channel's audio. This is very useful for building custom
979 vocoders or doing recording or analysis of the channel's audio in another
981 * Added a new module, res_config_curl, which permits using a HTTP POST url
982 to retrieve, create, update, and delete realtime information from a remote
983 web server. Note that this module requires func_curl.so to be loaded for
984 backend functionality.
985 * Added a new module, res_config_ldap, which permits the use of an LDAP
986 server for realtime data access.
987 * Added support for writing and running your dialplan in lua using the pbx_lua
988 module. See configs/extensions.lua.sample for examples of how to do this.
992 * Ability to use libcap to set high ToS bits when non-root
993 on Linux. If configure is unable to find libcap then you
994 can use --with-cap to specify the path.
995 * Added maxfiles option to options section of asterisk.conf which allows you to specify
996 what Asterisk should set as the maximum number of open files when it loads.
997 * Added the jittertargetextra configuration option.
998 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
999 configuration files for the IP channel drivers. The new option is "cos".
1000 This information is also documented in doc/qos.tex, or the IP Quality of Service
1001 section of asterisk.pdf.
1002 * When originating a call using AMI or pbx_spool that fails the reason for failure
1003 will now be available in the failed extension using the REASON dialplan variable.
1004 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
1005 It allows you to configure a prefix for auto-monitor recordings.
1006 * A new extension pattern matching algorithm, based on a trie, is introduced
1007 here, that could noticeably speed up mid-sized to large dialplans.
1008 It is NOT used by default, as duplicating the behaviour of the old pattern
1009 matcher is still under development. A config file option, in extensions.conf,
1010 in the [general] section, called "extenpatternmatchingnew", is by default
1011 set to false; setting that to true will force the use of the new algorithm.
1012 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
1013 be used to switch the algorithms at run time.
1014 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
1015 specifying which socket to use to connect to the running Asterisk daemon
1017 * Performance enhancements to the sched facility, which is used in
1018 the channel drivers, etc. Added hashtabs and doubly-linked lists
1019 to speed up deletion; start at the beginning or end of list to
1021 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
1022 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
1023 Added regression tests to the tests/ dir, also.
1024 * Added a refcount trace feature to astobj2 for those trying to balance
1025 object creation, deletion; work, play; space and time. See the
1026 notes in astobj2.h. Also, see utils/refcounter as well, as a
1027 quick way to find unbalanced refcounts in what could be a sea
1028 of objects that were balanced.
1029 * Added logging to 'make update' command. See update.log
1030 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
1031 do not come from the remote party.
1032 * Added the 'n' option to the SpeechBackground application to tell it to not
1033 answer the channel if it has not already been answered.
1034 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
1035 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
1037 * iLBC source code no longer included (see UPGRADE.txt for details)
1038 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
1039 deadlock is detected, a backtrace of the stack which led to the lock calls
1040 will be output to the CLI.
1041 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
1042 the "core show locks" CLI command will give lock information output as well
1043 as a backtrace of the stack which led to the lock calls.
1044 * users.conf now sports an optional alternateexts property, which permits
1045 allocation of additional extensions which will reach the specified user.
1046 * A new option for the configure script, --enable-internal-poll, has been added
1047 for use with systems which may have a buggy implementation of the poll system
1048 call. If you notice odd behavior such as the CLI being unresponsive on remote
1049 consoles, you may want to try using this option. This option is enabled by default
1050 on Darwin systems since it is known that the Darwin poll() implementation has
1054 --------------------
1055 * In addition to timing from DAHDI, there is a new timing module called
1056 res_timing_timerfd. In order to use this, you must be running Linux with
1057 a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
1058 script will be able to tell if you have the requirements. From menuselect, select
1059 res_timing_timerfd from the Resource Modules menu.
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