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 1.6.2 to Asterisk 1.8 ----------------
13 ------------------------------------------------------------------------------
17 * Added preferred_codec_only option in sip.conf. This feature limits the joint
18 codecs sent in response to an INVITE to the single most preferred codec.
19 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
20 to be used for the outgoing call. It must be one of the codecs configured
22 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
23 to be used for holding a private key. If tlsprivatekey is not specified,
24 tlscertfile is searched for both public and private key.
25 * Added tlsclientmethod option to sip.conf. This allows the protocol for
26 outbound client connections to be specified.
27 * The sendrpid parameter has been expanded to include the options
28 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
29 header to be sent (equivalent to setting sendrpid=yes) and setting
30 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
31 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
32 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
33 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
34 will accept the SDP even if the SDP version number is not properly incremented,
35 but will generate a warning in the log indicating that the SIP peer that sent
36 the SDP should have the 'ignoresdpversion' option set.
37 * The 'nat' option has now been been changed to have yes, no, force_rport, and
38 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
39 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
40 remote side requests it and disables symmetric RTP support. Setting it to
41 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
42 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
43 and enables symmetric RTP support.
44 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
45 response. This permits the master channel to know how each channel dialled
46 in a multi-channel setup resolved in an individual way.
47 * Added 'externtcpport' and 'externtlsport' options to allow custom port
48 configuration for the externip and externhost options when tcp or tls is used.
49 * Added support for message body (stored in content variable) to SIP NOTIFY message
50 accessible via AMI and CLI.
51 * Added 'media_address' configuration option which can be used to explicitly specify
52 the IP address to use in the SDP for media (audio, video, and text) streams.
53 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
54 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
56 * Added 'use_q850_reason' configuration option for generating and parsing
57 if available Reason: Q.850;cause=<cause code> header. It is implemented
58 in some gateways for better passing PRI/SS7 cause codes via SIP.
62 * Added rtsavesysname option into iax.conf to allow the systname to be saved
67 * Added ability to preset channel variables on indicated lines with the setvar
68 configuration option. Also, clearvars=all resets the list of variables back
70 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
71 See configs/res_pktccops.conf for more information.
75 * Added progress option to the app_dial D() option. When progress DTMF is
76 present, those values are sent immediately upon receiving a PROGRESS message
77 regardless if the call has been answered or not.
78 * Added functionality to the app_dial F() option to continue with execution
79 at the current location when no parameters are provided.
80 * Added the 'a' option to app_dial to answer the calling channel before any
81 announcements or macros are executed.
82 * Modified app_dial to set answertime when the called channel answers even if
83 the called channel hangs up during playback of an announcement.
84 * Added c() option to app_chanspy. This option allows custom DTMF to be set
85 to cycle through the next available channel. By default this is still '*'.
86 * Added x() option to app_chanspy. This option allows DTMF to be set to
88 * The Voicemail application has been improved to automatically ignore messages
89 that only contain silence.
90 * The ChanSpy application now has the 'S' option, which makes the application
91 automatically exit once it hits a point where no more channels are available
93 * The ChanSpy application also now has the 'E' option, which spies on a single
94 channel and exits when that channel hangs up.
95 * The MeetMe application now turns on the DENOISE() function by default, for
96 each participant. In our tests, this has significantly decreased background
97 noise (especially noisy data centers).
98 * Voicemail now permits storage of secrets in a separate file, located in the
99 spool directory of each individual user. The control for this is located in
100 the "passwordlocation" option in voicemail.conf. Please see the sample
101 configuration for more information.
102 * The ChanIsAvail application now exposes the returned cause code using a separate
103 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
104 * Added 'd' option to app_followme. This option disables the "Please hold"
106 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
107 received will terminate recording.
111 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
112 setting various connected line and redirecting party information.
113 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
114 support ISDN subaddressing.
115 * The CHANNEL() function now supports the "name" option.
116 * For DAHDI channels, the CHANNEL() dialplan function now
117 supports changing the channel's buffer policy (for the current
118 call only), using this syntax:
120 exten => s,n,Set(CHANNEL(buffers)=6,full)
122 This would change the channel to the 'full' buffer policy and
123 6 (six) buffers. Possible options for this setting are the same
124 as those in chan_dahdi.conf.
125 * For DAHDI channels, the CHANNEL() dialplan function now allows
126 the dialplan to request changes in the configuration of the active
127 echo canceller on the channel (if any), for the current call only.
130 exten => s,n,Set(CHANNEL(echocan_mode)=off)
132 The possible values are:
134 on - normal mode (the echo canceller is actually reinitialized)
136 fax - FAX/data mode (NLP disabled if possible, otherwise completely
138 voice - voice mode (returns from FAX mode, reverting the changes that
139 were made when FAX mode was requested)
140 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
141 and setting variables on the channel which created the current channel.
142 Administrators should take care to avoid naming conflicts, when multiple
143 channels are dialled at once, especially when used with the Local channel
144 construct (which all could set variables on the master channel). Usage
145 of the HASH() dialplan function, with the key set to the name of the slave
146 channel, is one approach that will avoid conflicts.
147 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
149 * func_odbc now allows multiple row results to be retrieved without using
150 mode=multirow. If rowlimit is set, then additional rows may be retrieved
151 from the same query by using the name of the function which retrieved the
152 first row as an argument to ODBC_FETCH().
153 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
154 dialplan. This function returns the content of the received message.
155 * Added REPLACE, which searches a given variable name for a set of characters,
156 then either replaces them with a single character or deletes them.
157 * Added PASSTHRU, which literally passes the same argument back as its return
158 value. The intent is to be able to use a literal string argument to
159 functions that currently require a variable name as an argument.
163 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
164 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
165 and is set when a dynamic feature is triggered.
169 * A new option, 'I' has been added to both app_queue and app_dial.
170 By setting this option, Asterisk will not update the caller with
171 connected line changes or redirecting party changes when they occur.
172 * A 'relative-peroidic-announce' option has been added to queues.conf. When
173 enabled, this option will cause periodic announce times to be calculated
174 from the end of announcements rather than from the beginning.
176 mISDN channel driver (chan_misdn) changes
177 ----------------------------------------
178 * Added display_connected parameter to misdn.conf to put a display string
179 in the CONNECT message containing the connected name and/or number if
180 the presentation setting permits it.
181 * Added display_setup parameter to misdn.conf to put a display string
182 in the SETUP message containing the caller name and/or number if the
183 presentation setting permits it.
184 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
185 indicate the dialplan settings are to be obtained from the asterisk
187 * Made misdn.conf parameter callerid accept the "name" <number> format
188 used by the rest of the system.
189 * Made use the nationalprefix and internationalprefix misdn.conf
190 parameters to prefix any received number from the ISDN link if that
191 number has the corresponding Type-Of-Number. NOTE: This includes
192 comparing the incoming call's dialed number against the MSN list.
193 * Added the following new parameters: unknownprefix, netspecificprefix,
194 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
195 received number from the ISDN link if that number has the corresponding
197 * Added new dialplan application misdn_command which permits controlling
198 the CCBS/CCNR functionality.
199 * Added new dialplan function mISDN_CC which permits retrieval of various
200 values from an active call completion record.
201 * For PTP, you should manually send the COLR of the redirected-to party
202 for an incomming redirected call if the incoming call could experience
203 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
204 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
205 if the REDIRECTING(from-num) is not empty.
206 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
207 option on all of the REDIRECTING statements before dialing the
208 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
209 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
210 redirecting-to presentation (COLR) when it becomes available.
211 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
214 thirdparty mISDN enhancements
215 -----------------------------
216 mISDN has been modified by Digium, Inc. to greatly expand facility message
218 * Enhanced COLP support for call diversion and transfer.
221 The latest modified mISDN v1.1.x based version is available at:
222 http://svn.digium.com/svn/thirdparty/mISDN/trunk
223 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
225 Tagged versions of the modified mISDN code are available under:
226 http://svn.digium.com/svn/thirdparty/mISDN/tags
227 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
229 libpri channel driver (chan_dahdi) DAHDI changes
230 -------------------------------------------
231 * The channel variable PRIREDIRECTREASON is now just a status variable
232 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
233 to read and alter the reason.
234 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
235 redirected-to party for an incomming redirected call if the incoming call
236 could experience further redirects. Just set the
237 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
238 to the COLR. A call has been redirected if the REDIRECTING(count) is not
240 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
241 use the inhibit(i) option on all of the REDIRECTING statements before
242 dialing the redirected-to party. You still have to set the
243 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
244 will update the redirecting-to presentation (COLR) when it becomes available.
245 * Added the ability to ignore calls that are not in a Multiple Subscriber
246 Number (MSN) list for PTMP CPE interfaces.
247 * Added dynamic range compression support for dahdi channels. It is
248 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
249 * Added support for ISDN calling and called subaddress with partial support
250 for connected line subaddress.
251 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
252 * Added handling of received HOLD/RETRIEVE messages and the optional ability
253 to transfer a held call on disconnect similar to an analog phone.
254 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
255 Will reroute/deflect an outgoing call when receive the message.
256 Can use the DAHDISendCallreroutingFacility to send the message for the
258 * Added standard location to add options to chan_dahdi dialing:
259 Dial(DAHDI/g1[/extension[/options]])
262 R Reverse charging indication
263 * Added Reverse Charging Indication (Collect calls) send/receive option.
264 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
265 Dial(DAHDI/g1/extension/R)
266 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
267 (requires latest LibPRI)
268 * Added ability to send/receive keypad digits in the SETUP message.
269 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
270 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
271 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
272 (requires latest LibPRI)
274 Asterisk Manager Interface
275 --------------------------
276 * The Hangup action now accepts a Cause header which may be used to
277 set the channel's hangup cause.
278 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
279 to specify a separate .pem file to hold a private key. By default sslcert
280 is used to hold both the public and private key.
281 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
282 for options containing the 'tls' prefix. For example, 'sslenable' is now
283 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
284 across all .conf files. All affected sample.conf files have been modified to
285 reflect this change. Previous options such as 'sslenable' still work,
286 but options with the 'tls' prefix are preferred.
287 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
288 in a channel. (res_mutestream.so)
289 * The configuration file manager.conf now supports a channelvars option, which
290 specifies a list of channel variables to include in each channel-oriented
293 Channel Event Logging
294 ---------------------
295 * A new interface, CEL, is introduced here. CEL logs single events, much like
296 the AMI, but it differs from the AMI in that it logs to db backends much
297 like CDR does; is based on the event subsystem introduced by Russell, and
298 can share in all its benefits; allows multiple backends to operate like CDR;
299 is specialized to event data that would be of concern to billing sytems,
300 like CDR. Backends for logging and accounting calls have been produced,
301 but a new CDR backend is still in development.
305 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR officianados.
306 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
307 etc are performed. Thus the peices of CDR can be grouped into multilegged sets.
308 * Multiple files and formats can now be specified in cdr_custom.conf.
309 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
310 See configs/cdr_syslog.conf.sample for more information.
311 * A 'sequence' field has been added to CDRs which can be combined with
312 linkedid or uniqueid to uniquely identify a CDR.
314 Calendaring for Asterisk
315 ------------------------
316 * A new set of modules were added supporing calendar integration with Asterisk.
317 Dialplan functions for reading from and writing to calendars are included,
318 as well as the ability to execute dialplan logic upon calendar event notifications.
319 iCalendar, CalDAV, and Exchange Server calendars are supported (Exchange support
320 only tested on Exchange Server 2003 with no support for forms-based authentication).
322 Multicast RTP Support
323 ---------------------
324 * A new RTP engine and channel driver have been added which supports Multicast RTP.
325 The channel driver can be used with the Page application to perform multicast RTP
326 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
327 Type can be either basic or linksys.
328 Destination is the IP address and port for the RTP packets.
329 Control address is specific to the linksys type and is used for sending the control
330 packets unique to them.
332 Security Events Framework
333 -------------------------
334 * Asterisk has a new C API for reporting security events. The module res_security_log
335 sends these events to the "security" logger level. Currently, AMI is the only
336 Asterisk component that reports security events. However, SIP support will be
337 coming soon. For more information on the security events framework, see the
338 "Security Events" chapter of the included documentation - doc/tex/asterisk.pdf.
342 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
343 XMPP text messages to the remote JID.
344 * Modules.conf has a new option - "require" - that marks a module as critical for
345 the execution of Asterisk.
346 If one of the required modules fail to load, Asterisk will exit with a return
348 * An 'X' option has been added to the asterisk application which enables #exec support.
349 This allows #exec to be used in asterisk.conf.
351 ------------------------------------------------------------------------------
352 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
353 ------------------------------------------------------------------------------
357 * The prematuremedia option is disabled by default. Applications requiring
358 SIP early audio must use the Progress() dialplan application to generate
359 the 183 progress message.
360 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
361 Snom phones use this for call pickup of extensions that the phone is
363 * Added support for subscribing to a voice mailbox on a remote server and
364 making the new/old message count available to local devices.
365 * Added support for setting the domain in the URI for caller of an
366 outbound call by using the SIPFROMDOMAIN channel variable.
367 * Added a new configuration option "remotesecret" for authentication to
368 remote services. For backwards compatibility, "secret" still has the
369 same function as before, but now you can configure both a remote secret and a
370 local secret for mutual authentication.
371 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
372 option is enabled, a SIP channel will go to the fax extension (if it exists)
373 after T38 is negotiated. This option is disabled by default.
374 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
375 the sound will be played to the target of an attended transfer
376 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
377 finer control over how many peers Asterisk will qualify and the gap between them
378 when all peers need to be qualified at the same time.
379 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
380 (either globally or for a specific peer), chan_sip will treat any SDP data
381 it receives as new data and update the media stream accordingly. By
382 default, Asterisk will only modify the media stream if the SDP session
383 version received is different from the current SDP session version. This
384 option is required to interoperate with devices that have non-standard SDP
385 session version implementations (observed with Microsoft OCS). This option
386 is disabled by default.
387 * The parsing of register => lines in sip.conf has been modified to allow a port
388 to be present in the "user" portion. Please see the sip.conf.sample file for more
390 * Added support for subscribing to MWI on a remote server and making the status available
391 as a mailbox. Please see the sip.conf.sample file for more information.
392 * Added a function to remove SIP headers added in the dialplan before the
393 first INVITE is generated - SIPRemoveHeader()
394 * Channel variables set with setvar= in a device configuration is now
395 set both for inbound and outbound calls.
396 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
400 * Added immediate option to iax.conf
401 * Added forceencryption option to iax.conf
402 * Added Encryption and Trunk status to manager command "iaxpeers"
406 * The configuration file now holds separate sections for devices and lines.
407 Please have a look at configs/skinny.conf.sample and change your skinny.conf
412 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
413 support for LibOpenR2. http://www.libopenr2.org/
414 * The UK option waitfordialtone has been added for use with BT analog
416 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
417 is used in conjunction with the 'faxdetect' configuration option. When
418 'faxbuffers' is used and fax tones are detected, the channel will dynamically
419 switch to the configured faxbuffers policy. For example, to use 6 buffers
420 and a 'full' buffer policy for a fax transmission, add:
422 The faxbuffers configuration will be in affect until the call is torn down.
423 * Added service message support for 4ESS/5ESS switches.
427 * Added a new dialplan function, CURLOPT, which permits setting various
428 options that may be useful with the CURL dialplan function, such as
429 cookies, proxies, connection timeouts, passwords, etc.
430 * Permit the syntax and synopsis fields of the corresponding dialplan
431 functions to be individually set from func_odbc.conf.
432 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
433 * func_odbc now may specify an insert query to execute, when the write query
434 affects 0 rows (usually indicating that no such row exists).
435 * Added a new dialplan function, LISTFILTER, which permits removing elements
436 from a set list, by name. Uses the same general syntax as the existing CUT
437 and FIELDQTY dialplan functions, which also manage lists.
438 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
439 obtaining realtime data from the dialplan.
440 * Added LOCAL_PEEK, which I have no idea how to use, but Leif Madsen wanted it.
441 Russell says it's, like, a scope resolution function for LOCAL variables.
442 Totally. Hopefully, that means more to you than it does to me.
443 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
444 of "core show function AUDIOHOOK_INHERIT" from the CLI
445 * Added AES_ENCRYPT. For information on its use, please see the output
446 of "core show function AES_ENCRYPT" from the CLI
447 * Added AES_DECRYPT. For information on its use, please see the output
448 of "core show function AES_DECRYPT" from the CLI
449 * func_odbc now supports database transactions across multiple queries.
453 * DAHDISendCallreroutingFacility parameters are now comma-separated,
454 instead of the old pipe.
455 * Scheduled meetme conferences may now have their end times extended by
457 * app_authenticate now gives the ability to select a prompt other than
459 * app_directory now pays attention to the searchcontexts setting in
460 voicemail.conf and will look through all contexts, if no context is
461 specified in the initial argument.
462 * A new application, Originate, has been introduced, that allows asynchronous
463 call origination from the dialplan.
464 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
465 in addition to the setting in the "general" context.
466 * Added ConfBridge dialplan application which does conference bridges without
467 DAHDI. For information on its use, please see the output of
468 "core show application ConfBridge" from the CLI.
472 * The Asterisk CLI has a new command, "channel redirect", which is similar in
473 operation to the AMI Redirect action.
474 * res_jabber: autoprune has been disabled by default, to avoid misconfiguration
475 that would end up being interpreted as a bug once Asterisk started removing
476 the contacts from a user list.
477 * extensions.conf now allows you to use keyword "same" to define an extension
478 without actually specifying an extension. It uses exactly the same pattern
479 as previously used on the last "exten" line. For example:
480 exten => 123,1,NoOp(something)
481 same => n,SomethingElse()
482 * musiconhold.conf classes of type 'files' can now use relative directory paths,
483 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
484 * All deprecated CLI commands are removed from the sourcecode. They are now handled
485 by the new clialiases module. See cli_aliases.conf.sample file.
486 * Times within timespecs are now accurate down to the minute. This is a change
487 from historical Asterisk, which only provided timespecs rounded to the nearest
488 even (read: evenly divisible by 2) minute mark.
489 * The realtime switch now supports an option flag, 'p', which disables searches for
491 * In addition to a time range and date range, timespecs now accept a 5th optional
492 argument, timezone. This allows you to perform time checks on alternate
493 timezones, especially if those daylight savings time ranges vary from your
494 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
496 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
497 give you the correct output for an asterisk box behind nat. It will give you the
498 externhost and localnet settings.
499 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
500 can connect calls in passthrough mode, as well as record and play back files.
501 * Successful and unsuccessful call pickup can now be alerted through sounds, by
502 using pickupsound and pickupfailsound in features.conf.
503 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
504 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
505 instead of the /var/run/asterisk.pid where it used to be. This will make
506 installs as non-root easier to manage.
508 Asterisk Manager Interface
509 --------------------------
510 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
511 a non-empty value) in your request. If you do this, any pending AMI events will
512 *not* be included in the response to your request as they would normally, but
513 will be left in the event queue for the next request you make to retrieve. For
514 some applications, this will allow you to guarantee that you will only see
515 events in responses to 'WaitEvent' actions, and can better know when to expect them.
516 To know whether the Asterisk server supports this header or not, your client can
517 inspect the first response back from the server to see if it includes this header:
519 Pragma: SuppressEvents
521 If this is included, the server supports event suppression.
523 * Added 4 new Actions to list skinny device(s) and line(s)
529 LDAP Schema File Additions
530 --------------------------
531 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
532 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
534 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
535 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
536 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
537 * Removed redundant IPaddr (there's already IPAddress)
538 - Gives more configuration Flags for SIP-Users available (tested)
539 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
540 without extensibleObject (which really should be the last resort); gives
541 also additional possibilities for LDAP-filter
543 ------------------------------------------------------------------------------
544 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
545 ------------------------------------------------------------------------------
547 Device State Handling
548 ---------------------
549 * The event infrastructure in Asterisk got another big update to help support
550 distributed events. It currently supports distributed device state and
551 distributed Voicemail MWI (Message Waiting Indication). A new module has
552 been merged, res_ais, which facilitates communicating events between servers.
553 It uses the SAForum AIS (Service Availability Forum Application Interface
554 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
555 a cluster of Asterisk servers, and to share events between them. For more
556 information on setting this up, see doc/distributed_devstate.txt.
560 * Added a new dialplan function, AST_CONFIG(), which allows you to access
561 variables from an Asterisk configuration file.
562 * The JACK_HOOK function now has a c() option to supply a custom client name.
563 * Added two new dialplan functions from libspeex for audio gain control and
564 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
565 rx directions of a channel from the dialplan.
566 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
567 based on other parameters. The default is still to search based on the
568 forwarding station ID. However, there are new options that allow you to search
569 based on the message desk terminal ID, or the message desk number.
570 * TIMEOUT() has been modified to be accurate down to the millisecond.
571 * ENUM*() functions now include the following new options:
572 - 'u' returns the full URI and does not strip off the URI-scheme.
573 - 's' triggers ISN specific rewriting
574 - 'i' looks for branches into an Infrastructure ENUM tree
575 - 'd' for a direct DNS lookup without any flipping of digits.
576 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
577 * CHANNEL() now has options for the maximum, minimum, and standard or normal
578 deviation of jitter, rtt, and loss for a call using chan_sip.
580 DAHDI channel driver (chan_dahdi) Changes
581 ----------------------------------------
582 * Channels can now be configured using named sections in chan_dahdi.conf, just
583 like other channel drivers, including the use of templates.
584 * The default for pridialplan has changed from 'national' to 'unknown'.
588 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
589 to something that matches the pattern a hint will be created using the contents
590 and variables evaluated.
591 * Dialplan matching has been extended to allow an extension to return to the
592 PBX core to wait for more digits. This is done by using the new dialplan
593 application called "Incomplete". This will permit a whole new level of
594 extension control, by giving the administrator more control over early
595 matches employing one of the short-circuit pattern match operators. Note
596 that custom applications can trigger this same behavior by returning the
597 special value AST_PBX_INCOMPLETE.
601 * Directory now permits both first and last names to be matched at the same
602 time. In addition, the number of digits to enter of the name can be set in
603 the arguments to Directory; previously, you could enter only 3, regardless
604 of how many names are in your company. For large companies, this should be
606 * Voicemail now permits a mailbox setting to wrap around from first to last
607 messages, if the "messagewrap" option is set to a true value.
608 * Voicemail now permits an external script to be run, for password validation.
609 The script should output "VALID" or "INVALID" on stdout, depending upon the
610 wish to validate or invalidate the password given. Arguments are:
611 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
613 * Dial has a new option: F(context^extension^pri), which permits a callee to
614 continue in the dialplan, at the specified label, if the caller hangs up.
615 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
616 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
617 * The Jack application now has a c() option to supply a custom client name.
618 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
619 like the pre-existing whisper mode, except that the spy can also talk to the
620 participant on the bridged channel as well.
621 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
622 to be spoken instead of the channel name or number. For more information on the
623 use of this option, issue the command "core show application ChanSpy" from the
625 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
626 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
627 words, if using the 'd' option, it is not possible to enter a number to append to
628 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
629 change to whisper mode, and pressing 6 will change to barge mode.
630 * ExternalIVR now takes several options that affect the way it performs, as
631 well as having several new commands. Please see doc/externalivr.txt for the
632 complete documentation.
633 * Added ability to communicate over a TCP socket instead of forking a child process for the
634 ExternalIVR application.
635 * ChanIsAvail has a new option, 'a', which will return all available channels instead
636 of just the first one if you give the function more then one channel to check.
637 * PrivacyManager now takes an option where you can specify a context where the
638 given number will be matched. This way you have more control over who is allowed
639 and it stops the people who blindly enter 10 digits.
640 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
641 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
642 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
643 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
644 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
645 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
646 * The Dial() application no longer copies the language used by the caller to the callee's
647 channel. If you desire for the caller's channel's language to be used for file playback
648 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
649 * SendImage() no longer hangs up the channel on error; instead, it sets the
650 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
651 'UNSUPPORTED'. This change makes SendImage() more consistent with other
653 * Park has a new option, 's', which silences the announcement of the parking space number.
654 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
655 invalid input and will be assumed to mean that no timeout is desired.
659 * Added DNS manager support to registrations for peers referencing peer entries.
660 DNS manager runs in the background which allows DNS lookups to be run asynchronously
661 as well as periodically updating the IP address. These properties allow for
662 better performance as well as recovery in the event of an IP change.
663 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
664 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
665 These changes also provide performance improvements for call setup and tear down.
666 * Added ability to specify registration expiry time on a per registration basis in
668 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
670 * Added t38pt_usertpsource option. See sip.conf.sample for details.
671 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
672 * 'sip show peers' and 'sip show users' display their entries sorted in
673 alphabetical order, as opposed to the order they were in, in the config
675 * Videosupport now supports an additional option, "always", which always sets
676 up video RTP ports, even on clients that don't support it. This helps with
677 callfiles and certain transfers to ensure that if two video phones are
678 connected, they will always share video feeds.
682 * Existing DNS manager lookups extended to check for SRV records.
683 * IAX2 encryption support has been improved to support periodic key rotation
684 within a call for enhanced security. The option "keyrotate" has been
685 provided to disable this functionality to preserve backwards compatibility
686 with older versions of IAX2 that do not support key rotation.
690 * New CLI command, "config reload <file.conf>" which reloads any module that
691 references that particular configuration file. Also added "config list"
692 which shows which configuration files are in use.
693 * New CLI commands, "pri show version" and "ss7 show version" that will
694 display which version of libpri and libss7 are being used, respectively.
695 A new API call was added so trunk will now have to be compiled against
696 a versions of libpri and libss7 that have them or it will not know that
697 these libraries exist.
698 * The commands "core show globals", "core set global" and "core set chanvar" has
699 been deprecated in favor of the more semanticly correct "dialplan show globals",
700 "dialplan set chanvar" and "dialplan set global".
701 * New CLI command "dialplan show chanvar" to list all variables associated
702 with a given channel.
706 * Addresses managed by DNS manager now can check to see if there is a DNS
707 SRV record for a given domain and will use that hostname/port if present.
709 AMI - The manager (TCP/TLS/HTTP)
710 --------------------------------
711 * The Status command now takes an optional list of variables to display
712 along with channel status.
713 * The QueueEntry event now also includes the channel's uniqueid
717 * res_odbc no longer has a limit of 1023 total possible unshared connections,
718 as some people were running into this limit. This limit has been increased
723 * The TRANSFER queue log entry now includes the the caller's original
724 position in the transferred-from queue.
725 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
726 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
727 as well as an explanation about timeout options in general
728 * Added a new option - C - for forcing the "answered elsewhere" flag on
729 cancellation of calls in to members of the queue. This is to avoid the
730 call to a member of a queue having the call listed as a "missed call".
734 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
735 adaptive capabilities. What this means in practical terms is that if your
736 realtime table lacks critical fields, Asterisk will now emit warnings to
737 that effect. Also, some of the realtime drivers have the ability (if
738 configured) to automatically add those columns to the table with the
739 correct type and length.
743 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
744 the 'setvar' option to cause a given audio file to be played upon completion
745 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
746 Skinny channels only.
747 * You can now compile Asterisk against the Hoard Memory Allocator, see doc/hoard.txt
748 for more information.
749 * Config file variables may now be appended to, by using the '+=' append
750 operator. This is most helpful when working with long SQL queries in
751 func_odbc.conf, as the queries no longer need to be specified on a single
753 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
754 which will add a second to the billsec when the ending
755 time is set, if the number in the microseconds field of the end time is
756 greater than the number of microseconds in the answer time. This allows
757 users to count the 'initiated' seconds in their billing records.
759 ------------------------------------------------------------------------------
760 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
761 ------------------------------------------------------------------------------
763 AMI - The manager (TCP/TLS/HTTP)
764 --------------------------------
765 * Manager has undergone a lot of changes, all of them documented
766 in doc/manager_1_1.txt
767 * Manager version has changed to 1.1
768 * Added a new action 'CoreShowChannels' to list currently defined channels
769 and some information about them.
770 * Added a new action 'SIPshowregistry' to list SIP registrations.
771 * Added TLS support for the manager interface and HTTP server
772 * Added the URI redirect option for the built-in HTTP server
773 * The output of CallerID in Manager events is now more consistent.
774 CallerIDNum is used for number and CallerIDName for name.
775 * Enable https support for builtin web server.
776 See configs/http.conf.sample for details.
777 * Added a new action, GetConfigJSON, which can return the contents of an
778 Asterisk configuration file in JSON format. This is intended to help
779 improve the performance of AJAX applications using the manager interface
781 * SIP and IAX manager events now use "ChannelType" in all cases where we
782 indicate channel driver. Previously, we used a mixture of "Channel"
783 and "ChannelDriver" headers.
784 * Added a "Bridge" action which allows you to bridge any two channels that
785 are currently active on the system.
786 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
787 the voicemail users setup.
788 * Added 'DBDel' and 'DBDelTree' manager commands.
789 * cdr_manager now reports events via the "cdr" level, separating it from
790 the very verbose "call" level.
791 * Manager users are now stored in memory. If you change the manager account
792 list (delete or add accounts) you need to reload manager.
793 * Added Masquerade manager event for when a masquerade happens between
795 * Added "manager reload" command for the CLI
796 * Lots of commands that only provided information are now allowed under the
797 Reporting privilege, instead of only under Call or System.
798 * The IAX* commands now require either System or Reporting privilege, to
799 mirror the privileges of the SIP* commands.
800 * Added ability to retrieve list of categories in a config file.
801 * Added ability to retrieve the content of a particular category.
802 * Added ability to empty a context.
803 * Created new action to create a new file.
804 * Updated delete action to allow deletion by line number with respect to category.
805 * Added new action insert to add new variable to category at specified line.
806 * Updated action newcat to allow new category to be inserted in file above another
808 * Added new event "JitterBufStats" in the IAX2 channel
809 * Originate now requires the Originate privilege and, if you want to call out
810 to a subshell, it requires the System privilege, as well. This was done to
811 enhance manager security.
812 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
813 * New command: Atxfer. See doc/manager_1_1.txt for more details or
814 manager show command Atxfer from the CLI
815 * New command: IAXregistry. See doc/manager_1_1.txt for more details or
816 manager show command IAXregistry from the CLI
820 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
821 state in the dialplan, as well as creating custom device states that are
822 controllable from the dialplan.
823 * Extend CALLERID() function with "pres" and "ton" parameters to
824 fetch string representation of calling number presentation indicator
825 and numeric representation of type of calling number value.
826 * MailboxExists converted to dialplan function
827 * A new option to Dial() for telling IP phones not to count the call
828 as "missed" when dial times out and cancels.
829 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
830 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
831 held for any given channel. Also, locks are automatically freed when a
833 * Added HINT() dialplan function that allows retrieving hint information.
834 Hints are mappings between extensions and devices for the sake of
835 determining the state of an extension. This function can retrieve the list
836 of devices or the name associated with a hint.
837 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
839 * Added SYSINFO() dialplan function which allows retrieval of system information
840 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
841 the existence of a dialplan target.
842 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
843 upper and lower case, respectively.
844 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
845 ID for the call (not the Asterisk call ID or unique ID), provided that the
846 channel driver supports this. For SIP, you get the SIP call-ID for the
847 bridged channel which you can store in the CDR with a custom field.
851 * Added CLI permissions, config file: cli_permissions.conf
852 default is to allow all commands for every local user/group.
853 Also this new feature added three new CLI commands:
854 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
855 - cli reload permissions
856 - cli show permissions
857 * New CLI command "core show hint" (usage: core show hint <exten>)
858 * New CLI command "core show settings"
859 * Added 'core show channels count' CLI command.
860 * Added the ability to set the core debug and verbose values on a per-file basis.
861 * Added 'queue pause member' and 'queue unpause member' CLI commands
862 * Ability to set process limits ("ulimit") without restarting Asterisk
863 * Enhanced "agi debug" to print the channel name as a prefix to the debug
864 output to make debugging on busy systems much easier.
865 * New CLI commands "dialplan set extenpatternmatching true/false"
866 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
867 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
868 listed in the startup_commands section of cli.conf will get executed.
869 * Added a CLI command, "devstate change", which allows you to set custom device
870 states from the func_devstate module that provides the DEVICE_STATE() function
871 and handling of the "Custom:" devices.
872 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
873 sorted into the different possible callbacks, with the number of entries
874 currently scheduled for each. Gives you a feel for how busy the sip channel
876 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
877 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
878 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
882 * Improved NAT and STUN support.
883 chan_sip now can use port numbers in bindaddr, externip and externhost
884 options, as well as contact a STUN server to detect its external address
885 for the SIP socket. See sip.conf.sample, 'NAT' section.
886 * The default SIP useragent= identifier now includes the Asterisk version
887 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
888 If set, and the incoming request carries authentication info,
889 the username to match in the users list is taken from the Digest header
890 rather than from the From: field. This feature is considered experimental.
891 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
892 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
893 * The "localmask" setting was removed in version 1.2 and the reminder about it
894 being removed is now also removed.
895 * A new option "busylevel" for setting a level of calls where asterisk reports
896 a device as busy, to separate it from call-limit. This value is also added
897 to the SIP_PEER dialplan function.
898 * A new realtime family called "sipregs" is now supported to store SIP registration
899 data. If this family is defined, "sippeers" will be used for configuration and
900 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
901 registration data, as before.
902 * The SIPPEER function have new options for port address, call and pickup groups
903 * Added support for T.140 realtime text in SIP/RTP
904 * The "checkmwi" option has been removed from sip.conf, as it is no longer
905 required due to the restructuring of how MWI is handled. See the descriptions
906 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
907 for more information.
908 * Added rtpdest option to CHANNEL() dialplan function.
909 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
910 * SIP now adds a header to the CANCEL if the call was answered by another phone
911 in the same dial command, or if the new c option in dial() is used.
912 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
913 states it is not needed. For phones, however, that do require it the "registertrying" option
914 has been added so it can be enabled.
915 * A new option called "callcounter" (global/peer/user level) enables call counters needed
916 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
917 used to enable this functionality).
918 * New settings for timer T1 and timer B on a global level or per device. This makes it
919 possible to force timeout faster on non-responsive SIP servers. These settings are
920 considered advanced, so don't use them unless you have a problem.
921 * Added a dial string option to be able to set the To: header in an INVITE to any
923 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
924 the qualify frequency.
925 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
926 were not properly torn down due to network or endpoint failures during an established
928 * Added experimental TCP and TLS support for SIP. See doc/siptls.txt and
929 configs/sip.conf.sample for more information on how it is used.
930 * Added a new configuration option "authfailureevents" that enables manager events when
931 a peer can't authenticate properly.
932 * Added DNS manager support to registrations for peers not referencing a peer entry.
936 * Added the trunkmaxsize configuration option to chan_iax2.
937 * Added the srvlookup option to iax.conf
938 * Added support for OSP. The token is set and retrieved through the CHANNEL()
941 XMPP Google Talk/Jingle changes
942 -------------------------------
943 * Added the bindaddr option to gtalk.conf.
947 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
948 * Proper codec support in chan_skinny.
949 * Added settings for IP and Ethernet QoS requests
953 * Added separate settings for media QoS in mgcp.conf
955 Console Channel Driver changes
956 ------------------------------
957 * Added experimental support for video send & receive to chan_oss.
958 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
961 Phone channel changes (chan_phone)
962 ----------------------------------
963 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
965 H.323 channel Changes
966 ---------------------
967 * H323 remote hold notification support added (by NOTIFY message
968 and/or H.450 supplementary service)
970 Local channel changes
971 ---------------------
972 * The device state functionality in the Local channel driver has been updated
973 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
974 to just UNKNOWN if the extension exists.
975 * Added jitterbuffer support for chan_local. This allows you to use the
976 generic jitterbuffer on incoming calls going to Asterisk applications.
977 For example, this would allow you to use a jitterbuffer for an incoming
978 SIP call to Voicemail by putting a Local channel in the middle. This
979 feature is enabled by using the 'j' option in the Dial string to the Local
980 channel in conjunction with the existing 'n' option for local channels.
981 * A 'b' option has been added which causes chan_local to return the actual channel
982 that is behind it when queried. This is useful for transfer scenarios as the
983 actual channel will be transferred, not the Local channel.
985 Agent channel changes
986 ----------------------
987 * The ackcall and endcall options are now supplemented with options acceptdtmf
988 and enddtmf. These allow for the DTMF keypress to be configurable. The options
989 default to their old hard-coded values ('#' and '*' respectively) so this should
990 not break any existing agent installations.
992 DAHDI channel driver (chan_dahdi) Changes
993 ----------------------------------------
994 * SS7 support (via libss7 library)
995 * In India, some carriers transmit CID via dtmf. Some code has been added
996 that will handle some situations. The cidstart=polarity_IN choice has been added for
997 those carriers that transmit CID via dtmf after a polarity change.
998 * CID matching information is now shown when doing 'dialplan show'.
999 * Added dahdi show version CLI command.
1000 * Added setvar support to chan_dahdi.conf channel entries.
1001 * Added two new options: mwimonitor and mwimonitornotify. These options allow
1002 you to enable MWI monitoring on FXO lines. When the MWI state changes,
1003 the script specified in the mwimonitornotify option is executed. An internal
1004 event indicating the new state of the mailbox is also generated, so that
1005 the normal MWI facilities in Asterisk work as usual.
1006 * Added signalling type 'auto', which attempts to use the same signalling type
1007 for a channel as configured in DAHDI. This is primarily designed for analog
1008 ports, but will also work for digital ports that are configured for FXS or FXO
1009 signalling types. This mode is also the default now, so if your chan_dahdi.conf
1010 does not specify signalling for a channel (which is unlikely as the sample
1011 configuration file has always recommended specifying it for every channel) then
1012 the 'auto' mode will be used for that channel if possible.
1013 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
1014 state for a channel; also ensured that the DNDState Manager event is
1015 emitted no matter how the DND state is set or cleared.
1019 * Added a new channel driver, chan_unistim. See doc/unistim.txt and
1020 configs/unistim.conf.sample for details. This new channel driver allows
1021 you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
1022 * Added a new channel driver, chan_console, which uses portaudio as a cross
1023 platform audio interface. It was written as a channel driver that would
1024 work with Mac CoreAudio, but portaudio supports a number of other audio
1025 interfaces, as well. Note that this channel driver requires v19 or higher
1026 of portaudio; older versions have a different API.
1030 * Added the ability to specify arguments to the Dial application when using
1031 the DUNDi switch in the dialplan.
1032 * Added the ability to set weights for responses dynamically. This can be
1033 done using a global variable or a dialplan function. Using the SHELL()
1034 function would allow you to have an external script set the weight for
1036 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
1037 functions will allow you to initiate a DUNDi query from the dialplan,
1038 find out how many results there are, and access each one.
1042 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
1043 functions will allow you to initiate an ENUM lookup from the dialplan,
1044 and Asterisk will cache the results. ENUMRESULT can be used to access
1045 the results without doing multiple DNS queries.
1049 * Added the ability to customize which sound files are used for some of the
1050 prompts within the Voicemail application by changing them in voicemail.conf
1051 * Added the ability for the "voicemail show users" CLI command to show users
1052 configured by the dynamic realtime configuration method.
1053 * MWI (Message Waiting Indication) handling has been significantly
1054 restructured internally to Asterisk. It is now totally event based
1055 instead of polling based. The voicemail application will notify other
1056 modules that have subscribed to MWI events when something in the mailbox
1058 This also means that if any other entity outside of Asterisk is changing
1059 the contents of mailboxes, then the voicemail application still needs to
1060 poll for changes. Examples of situations that would require this option
1061 are web interfaces to voicemail or an email client in the case of using
1062 IMAP storage. So, two new options have been added to voicemail.conf
1063 to account for this: "pollmailboxes" and "pollfreq". See the sample
1064 configuration file for details.
1065 * Added "tw" language support
1066 * Added support for storage of greetings using an IMAP server
1067 * Added ability to customize forward, reverse, stop, and pause keys for message playback
1068 * SMDI is now enabled in voicemail using the smdienable option.
1069 * A "lockmode" option has been added to asterisk.conf to configure the file
1070 locking method used for voicemail, and potentially other things in the
1071 future. The default is the old behavior, lockfile. However, there is a
1072 new method, "flock", that uses a different method for situations where the
1073 lockfile will not work, such as on SMB/CIFS mounts.
1074 * Added the ability to backup deleted messages, to ease recovery in the case
1075 that a user accidentally deletes a message, and discovers that they need it.
1076 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
1077 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
1078 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
1079 voicemail boxes. The SMDI interface can also poll for MWI changes when some
1080 outside entity is modifying the state of the mailbox (such as IMAP storage or
1081 a web interface of some kind).
1082 * Added the support for marking messages as "urgent." There are two methods to accomplish
1083 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
1084 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
1085 the message as urgent after he has recorded a voicemail by following the voice instructions.
1086 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
1091 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
1092 used across multiple queues.
1093 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
1094 setqueueentryvar options for each queue, see queues.conf.sample for details.
1095 * Added keepstats option to queues.conf which will keep queue
1096 statistics during a reload.
1097 * setinterfacevar option in queues.conf also now sets a variable
1098 called MEMBERNAME which contains the member's name.
1099 * Added 'Strategy' field to manager event QueueParams which represents
1100 the queue strategy in use.
1101 * Added option to run macro when a queue member is connected to a caller,
1102 see queues.conf.sample for details.
1103 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
1104 does not count paused queue members as unavailable.
1105 * Added min-announce-frequency option to queues.conf which allows you to control the
1106 minimum amount of time between queue announcements for use when the caller's queue
1107 position changes frequently.
1108 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
1110 * Added ability for non-realtime queues to have realtime members
1111 * Added the "linear" strategy to queues.
1112 * Added the "wrandom" strategy to queues.
1113 * Added new channel variable QUEUE_MIN_PENALTY
1114 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
1115 rules in queuerules.conf. See configs/queuerules.conf.sample for details
1116 * Added a new parameter for member definition, called state_interface. This may be
1117 used so that a member may be called via one interface but have a different interface's
1118 device state reported.
1119 * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
1120 "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
1121 "manager show command QueueReset."
1122 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
1123 specified by the periodic-announce option, then one will be chosen randomly when it is time
1124 to play a periodic announcment
1125 * New configuration options: announce-position now takes two more values in addition to "yes" and
1126 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
1127 announce-position-limit. By setting announce-position to "limit" callers will only have their
1128 position announced if their position is less than what is specified by announce-position-limit.
1129 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
1130 will be told that their are more than announce-position-limit callers waiting.
1131 * Two new queue log events have been added. An ADDMEMBER event will be logged
1132 when a realtime queue member is added and a REMOVEMEMBER event will be logged
1133 when a realtime queue member is removed. Since there is no calling channel associated
1134 with these events, the string "REALTIME" is placed where the channel's unique id
1135 is typically placed.
1136 * The configuration method for the "joinempty" and "leavewhenempty" options has
1137 changed to a comma-separated list of methods of determining member availability
1138 instead of vague terms such as "yes," "loose," "no," and "strict." These old four
1139 values are still accepted for backwards-compatibility, though.
1140 * The average talktime is now calculated on queues. This information is reported via the
1141 CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
1142 and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
1147 * The 'o' option to provide an optimization has been removed and its functionality
1148 has been enabled by default.
1149 * When a conference is created, the UNIQUEID of the channel that caused it to be
1150 created is stored. Then, every channel that joins the conference will have the
1151 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
1152 callers that come and go from long standing conferences.
1153 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
1154 except it does operations on a channel by name, instead of number in a conference.
1155 This is a very useful feature in combination with the 'X' option to ChanSpy.
1156 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
1158 * Added new RealTime functionality to provide support for scheduled conferencing.
1159 This includes optional messages to the caller if they attempt to join before
1160 the schedule start time, or to allow the caller to join the conference early.
1161 Also included is optional support for limiting the number of callers per
1162 RealTime conference.
1163 * Added the S() and L() options to the MeetMe application. These are pretty
1164 much identical to the S() and L() options to Dial(). They let you set
1165 timeouts for the conference, as well as have warning sounds played to
1166 let the caller know how much time is left, and when it is running out.
1167 * Added the ability to do "meetme concise" with the "meetme" CLI command.
1168 This extends the concise capabilities of this CLI command to include
1169 listing all conferences, instead of an addition to the other sub commands
1170 for the "meetme" command.
1171 * Added the ability to specify the music on hold class used to play into the
1172 conference when there is only one member and the M option is used.
1173 * Added MEETME_INFO dialplan function which provides a way to query
1174 various properties of a Meetme conference.
1176 Other Dialplan Application Changes
1177 ----------------------------------
1178 * Argument support for Gosub application
1179 * From the to-do lists: straighten out the app timeout args:
1180 Wait() app now really does 0.3 seconds- was truncating arg to an int.
1181 WaitExten() same as Wait().
1182 Congestion() - Now takes floating pt. argument.
1183 Busy() - now takes floating pt. argument.
1184 Read() - timeout now can be floating pt.
1185 WaitForRing() now takes floating pt timeout arg.
1186 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
1187 * Added 's' option to Page application.
1188 * Added an optional timeout argument to the Page application.
1189 * Added 'E', 'V', and 'P' commands to ExternalIVR.
1190 * Added 'o' and 'X' options to Chanspy.
1191 * Added a new dialplan application, Bridge, which allows you to bridge the
1192 calling channel to any other active channel on the system.
1193 * Added the ability to specify a music on hold class to play instead of ringing
1194 for the SLATrunk application.
1195 * The Read application no longer exits the dialplan on error. Instead, it sets
1196 READSTATUS to ERROR, which you can catch and handle separately.
1197 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
1198 of asking for verification of each name, one at a time.
1199 * Privacy() no longer uses privacy.conf, as all options are specifyable as
1200 direct options to the app.
1201 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
1203 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
1204 * The ChannelRedirect application no longer exits the dialplan if the given channel
1205 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
1206 or NOCHANNEL if the given channel was not found.
1207 * The silencethreshold setting that was previously configurable in multiple
1208 applications is now settable globally via dsp.conf.
1210 Music On Hold Changes
1211 ---------------------
1212 * A new option, "digit", has been added for music on hold classes in
1213 musiconhold.conf. If this is set for a music on hold class, a caller
1214 listening to music on hold can press this digit to switch to listening
1215 to this music on hold class.
1216 * Support for realtime music on hold has been added.
1217 * In conjunction with the realtime music on hold, a general section has
1218 been added to musiconhold.conf, its sole variable is cachertclasses. If this
1219 is set, then music on hold classes found in realtime will be cached in memory.
1223 * AEL upgraded to use the Gosub with Arguments instead
1224 of Macro application, to hopefully reduce the problems
1225 seen with the artificially low stack ceiling that
1226 Macro bumps into. Macros can only call other Macros
1227 to a depth of 7. Tests run using gosub, show depths
1228 limited only by virtual memory. A small test demonstrated
1229 recursive call depths of 100,000 without problems.
1230 -- in addition to this, all apps that allowed a macro
1231 to be called, as in Dial, queues, etc, are now allowing
1232 a gosub call in similar fashion.
1233 * AEL now generates LOCAL(argname) declarations when it
1234 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
1235 etc. That makes the arguments local in scope. The user
1236 can define their own local variables in macros, now,
1237 by saying "local myvar=someval;" or using Set() in this
1238 fashion: Set(LOCAL(myvar)=someval); ("local" is now
1240 * utils/conf2ael introduced. Will convert an extensions.conf
1241 file into extensions.ael. Very crude and unfinished, but
1242 will be improved as time goes by. Should be useful for a
1243 first pass at conversion.
1244 * aelparse will now read extensions.conf to see if a referenced
1245 macro or context is there before issueing a warning.
1246 * AEL parser sets a local channel variable ~~EXTEN~~, to
1247 preserve the value of ${EXTEN} thru switch statements.
1248 * New operator in $[...] expressions: the ~~ operator serves
1249 as a concatenation operator. AT THE MOMENT, it is really only
1250 necessary and useful in AEL, especially in if() expressions.
1251 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
1252 any enclosing double-quotes, and evaluate to the value of a
1253 concatenated with the value of b. For example if a is set to
1254 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
1255 evaluate to xyzabc .
1258 Call Features (res_features) Changes
1259 ------------------------------------
1260 * Added the parkedcalltransfers option to features.conf
1261 * Added parkedcallparking option to control one touch parking w/ parking
1263 * Added parkedcallhangup option to control disconnect feature w/ parking
1265 * Added parkedcallrecording option to control one-touch record w/ parking
1267 * Added BRIDGE_FEATURES variable to set available features for a channel
1268 * The built-in method for doing attended transfers has been updated to
1269 include some new options that allow you to have the transferee sent
1270 back to the person that did the transfer if the transfer is not successful.
1271 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
1272 in features.conf.sample.
1273 * Added support for configuring named groups of custom call features in
1274 features.conf. This means that features can be written a single time, and
1275 then mapped into groups of features for different key mappings or easier
1277 * Updated the ParkedCall application to allow you to not specify a parking
1278 extension. If you don't specify a parking space to pick up, it will grab
1279 the first one available.
1280 * Added cli command 'features reload' to reload call features from features.conf
1281 * Moved into core asterisk binary.
1282 * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
1284 Language Support Changes
1285 ------------------------
1286 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
1287 * Added support for the Hungarian language for saying numbers, dates, and times.
1291 * Added SPEECH commands for speech recognition. A complete listing can be found
1293 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
1294 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
1295 does not behave as expected; the native command needs to be used, instead.
1299 * Added rotatestrategy option to logger.conf, along with two new options:
1300 "timestamp" which will use the time to name the logger files instead of
1301 sequence number; and "rotate", which rotates the names of the log files,
1302 similar to the way syslog rotates files.
1303 * Added exec_after_rotate option to logger.conf, which allows a system
1304 command to be run after rotation. This is primarily useful with
1305 rotatestrategy=rotate, to allow a limit on the number of log files kept
1306 and to ensure that the oldest log file gets deleted.
1307 * Added realtime support for the queue log
1311 * The cdr_manager module has a [mappings] feature, like cdr_custom,
1312 to add fields to the manager event from the CDR variables.
1313 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
1314 backend database CDR table. Specifically, additional, non-standard
1315 columns are supported, merely by setting the corresponding CDR variable in
1316 your dialplan. In addition, you may alias any column to another name (for
1317 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
1318 simply "alias src => ANI" in the configuration file). Records may be
1319 posted to more than one backend, simply by specifying multiple categories
1320 in the configuration file. And finally, you may filter which CDRs get
1321 posted to each backend, by specifying a filter (which the record must
1322 match) for the particular category. Filters are additive (meaning all
1323 rules must match to post that CDR).
1324 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
1325 module. Specifically, you may add additional columns into the table and
1326 they will be set, if you set the corresponding CDR variable name. Also,
1327 if you omit columns in your database table, they will be silently skipped
1328 (but a record will still be inserted, based on what columns remain). Note
1329 that the other two features from cdr_adaptive_odbc (alias and filter) are
1330 not currently supported.
1331 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
1332 has been disabled using the NoCDR application.
1334 Miscellaneous New Modules
1335 -------------------------
1336 * Added a new CDR module, cdr_sqlite3_custom.
1337 * Added a new realtime configuration module, res_config_sqlite
1338 * Added a new codec translation module, codec_resample, which re-samples
1339 signed linear audio between 8 kHz and 16 kHz to help support wideband
1341 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
1342 based on configuration templates that use Asterisk dialplan function and
1343 variable substitution. It should be possible to create phone profiles and
1344 templates that work for the majority of phones provisioned over http. It
1345 is currently only intended to provision a single user account per phone.
1346 An example profile and set of templates for Polycom phones is provided.
1347 NOTE: Polycom firmware is not included, but should be placed in
1348 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
1349 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
1350 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
1351 provided; there is a JACK() application, and a JACK_HOOK() function. Both
1352 interfaces create an input and output JACK port. The application makes
1353 these ports the endpoint of the call. The audio coming from the channel
1354 goes out the output port and whatever comes back in on the input port is
1355 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
1356 audiohook on the channel. This lets you run the audio coming from a
1357 channel through JACK, and whatever comes back in is what gets forwarded
1358 on as the channel's audio. This is very useful for building custom
1359 vocoders or doing recording or analysis of the channel's audio in another
1361 * Added a new module, res_config_curl, which permits using a HTTP POST url
1362 to retrieve, create, update, and delete realtime information from a remote
1363 web server. Note that this module requires func_curl.so to be loaded for
1364 backend functionality.
1365 * Added a new module, res_config_ldap, which permits the use of an LDAP
1366 server for realtime data access.
1367 * Added support for writing and running your dialplan in lua using the pbx_lua
1368 module. See configs/extensions.lua.sample for examples of how to do this.
1372 * Ability to use libcap to set high ToS bits when non-root
1373 on Linux. If configure is unable to find libcap then you
1374 can use --with-cap to specify the path.
1375 * Added maxfiles option to options section of asterisk.conf which allows you to specify
1376 what Asterisk should set as the maximum number of open files when it loads.
1377 * Added the jittertargetextra configuration option.
1378 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
1379 configuration files for the IP channel drivers. The new option is "cos".
1380 This information is also documented in doc/qos.tex, or the IP Quality of Service
1381 section of asterisk.pdf.
1382 * When originating a call using AMI or pbx_spool that fails the reason for failure
1383 will now be available in the failed extension using the REASON dialplan variable.
1384 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
1385 It allows you to configure a prefix for auto-monitor recordings.
1386 * A new extension pattern matching algorithm, based on a trie, is introduced
1387 here, that could noticeably speed up mid-sized to large dialplans.
1388 It is NOT used by default, as duplicating the behaviour of the old pattern
1389 matcher is still under development. A config file option, in extensions.conf,
1390 in the [general] section, called "extenpatternmatchingnew", is by default
1391 set to false; setting that to true will force the use of the new algorithm.
1392 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
1393 be used to switch the algorithms at run time.
1394 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
1395 specifying which socket to use to connect to the running Asterisk daemon
1397 * Performance enhancements to the sched facility, which is used in
1398 the channel drivers, etc. Added hashtabs and doubly-linked lists
1399 to speed up deletion; start at the beginning or end of list to
1401 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
1402 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
1403 Added regression tests to the tests/ dir, also.
1404 * Added a refcount trace feature to astobj2 for those trying to balance
1405 object creation, deletion; work, play; space and time. See the
1406 notes in astobj2.h. Also, see utils/refcounter as well, as a
1407 quick way to find unbalanced refcounts in what could be a sea
1408 of objects that were balanced.
1409 * Added logging to 'make update' command. See update.log
1410 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
1411 do not come from the remote party.
1412 * Added the 'n' option to the SpeechBackground application to tell it to not
1413 answer the channel if it has not already been answered.
1414 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
1415 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
1417 * iLBC source code no longer included (see UPGRADE.txt for details)
1418 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
1419 deadlock is detected, a backtrace of the stack which led to the lock calls
1420 will be output to the CLI.
1421 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
1422 the "core show locks" CLI command will give lock information output as well
1423 as a backtrace of the stack which led to the lock calls.
1424 * users.conf now sports an optional alternateexts property, which permits
1425 allocation of additional extensions which will reach the specified user.
1426 * A new option for the configure script, --enable-internal-poll, has been added
1427 for use with systems which may have a buggy implementation of the poll system
1428 call. If you notice odd behavior such as the CLI being unresponsive on remote
1429 consoles, you may want to try using this option. This option is enabled by default
1430 on Darwin systems since it is known that the Darwin poll() implementation has
1434 --------------------
1435 * In addition to timing from DAHDI, there is a new timing module called
1436 res_timing_timerfd. In order to use this, you must be running Linux with
1437 a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
1438 script will be able to tell if you have the requirements. From menuselect, select
1439 res_timing_timerfd from the Resource Modules menu.
projects / asterisk/asterisk.git / blob