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 .m3u support for Mp3Player application.
76 * Added progress option to the app_dial D() option. When progress DTMF is
77 present, those values are sent immediately upon receiving a PROGRESS message
78 regardless if the call has been answered or not.
79 * Added functionality to the app_dial F() option to continue with execution
80 at the current location when no parameters are provided.
81 * Added the 'a' option to app_dial to answer the calling channel before any
82 announcements or macros are executed.
83 * Modified app_dial to set answertime when the called channel answers even if
84 the called channel hangs up during playback of an announcement.
85 * Added c() option to app_chanspy. This option allows custom DTMF to be set
86 to cycle through the next available channel. By default this is still '*'.
87 * Added x() option to app_chanspy. This option allows DTMF to be set to
89 * The Voicemail application has been improved to automatically ignore messages
90 that only contain silence.
91 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
92 associated mailbox(es) to be greetings-only.
93 * The ChanSpy application now has the 'S' option, which makes the application
94 automatically exit once it hits a point where no more channels are available
96 * The ChanSpy application also now has the 'E' option, which spies on a single
97 channel and exits when that channel hangs up.
98 * The MeetMe application now turns on the DENOISE() function by default, for
99 each participant. In our tests, this has significantly decreased background
100 noise (especially noisy data centers).
101 * Voicemail now permits storage of secrets in a separate file, located in the
102 spool directory of each individual user. The control for this is located in
103 the "passwordlocation" option in voicemail.conf. Please see the sample
104 configuration for more information.
105 * The ChanIsAvail application now exposes the returned cause code using a separate
106 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
107 * Added 'd' option to app_followme. This option disables the "Please hold"
109 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
110 received will terminate recording.
111 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
112 Previously the folder could only be set per context, but has now been extended
113 using the imapfolder option.
114 * Voicemail now allows the pager date format to be specified separately from the
116 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
117 to allow joining, leaving, and sending text to group chats.
118 * MeetMe has a new option 'G' to play an announcement before joining a conference.
119 * Page has a new option 'A(x)' which will playback an announcement simultaneously
120 to all paged phones (and optionally excluding the caller's one using the new
121 option 'n') before the call is bridged.
125 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
126 setting various connected line and redirecting party information.
127 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
128 support ISDN subaddressing.
129 * The CHANNEL() function now supports the "name" option.
130 * For DAHDI channels, the CHANNEL() dialplan function now
131 supports changing the channel's buffer policy (for the current
132 call only), using this syntax:
134 exten => s,n,Set(CHANNEL(buffers)=6,full)
136 This would change the channel to the 'full' buffer policy and
137 6 (six) buffers. Possible options for this setting are the same
138 as those in chan_dahdi.conf.
139 * For DAHDI channels, the CHANNEL() dialplan function now allows
140 the dialplan to request changes in the configuration of the active
141 echo canceller on the channel (if any), for the current call only.
144 exten => s,n,Set(CHANNEL(echocan_mode)=off)
146 The possible values are:
148 on - normal mode (the echo canceller is actually reinitialized)
150 fax - FAX/data mode (NLP disabled if possible, otherwise completely
152 voice - voice mode (returns from FAX mode, reverting the changes that
153 were made when FAX mode was requested)
154 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
155 and setting variables on the channel which created the current channel.
156 Administrators should take care to avoid naming conflicts, when multiple
157 channels are dialled at once, especially when used with the Local channel
158 construct (which all could set variables on the master channel). Usage
159 of the HASH() dialplan function, with the key set to the name of the slave
160 channel, is one approach that will avoid conflicts.
161 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
163 * func_odbc now allows multiple row results to be retrieved without using
164 mode=multirow. If rowlimit is set, then additional rows may be retrieved
165 from the same query by using the name of the function which retrieved the
166 first row as an argument to ODBC_FETCH().
167 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
168 dialplan. This function returns the content of the received message.
169 * Added REPLACE, which searches a given variable name for a set of characters,
170 then either replaces them with a single character or deletes them.
171 * Added PASSTHRU, which literally passes the same argument back as its return
172 value. The intent is to be able to use a literal string argument to
173 functions that currently require a variable name as an argument.
177 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
178 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
179 and is set when a dynamic feature is triggered.
183 * A new config option, penaltymemberslimit, has been added to queues.conf.
184 When set this option will disregard penalty settings when a queue has too
186 * A new option, 'I' has been added to both app_queue and app_dial.
187 By setting this option, Asterisk will not update the caller with
188 connected line changes or redirecting party changes when they occur.
189 * A 'relative-peroidic-announce' option has been added to queues.conf. When
190 enabled, this option will cause periodic announce times to be calculated
191 from the end of announcements rather than from the beginning.
193 mISDN channel driver (chan_misdn) changes
194 ----------------------------------------
195 * Added display_connected parameter to misdn.conf to put a display string
196 in the CONNECT message containing the connected name and/or number if
197 the presentation setting permits it.
198 * Added display_setup parameter to misdn.conf to put a display string
199 in the SETUP message containing the caller name and/or number if the
200 presentation setting permits it.
201 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
202 indicate the dialplan settings are to be obtained from the asterisk
204 * Made misdn.conf parameter callerid accept the "name" <number> format
205 used by the rest of the system.
206 * Made use the nationalprefix and internationalprefix misdn.conf
207 parameters to prefix any received number from the ISDN link if that
208 number has the corresponding Type-Of-Number. NOTE: This includes
209 comparing the incoming call's dialed number against the MSN list.
210 * Added the following new parameters: unknownprefix, netspecificprefix,
211 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
212 received number from the ISDN link if that number has the corresponding
214 * Added new dialplan application misdn_command which permits controlling
215 the CCBS/CCNR functionality.
216 * Added new dialplan function mISDN_CC which permits retrieval of various
217 values from an active call completion record.
218 * For PTP, you should manually send the COLR of the redirected-to party
219 for an incomming redirected call if the incoming call could experience
220 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
221 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
222 if the REDIRECTING(from-num) is not empty.
223 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
224 option on all of the REDIRECTING statements before dialing the
225 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
226 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
227 redirecting-to presentation (COLR) when it becomes available.
228 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
231 thirdparty mISDN enhancements
232 -----------------------------
233 mISDN has been modified by Digium, Inc. to greatly expand facility message
235 * Enhanced COLP support for call diversion and transfer.
238 The latest modified mISDN v1.1.x based version is available at:
239 http://svn.digium.com/svn/thirdparty/mISDN/trunk
240 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
242 Tagged versions of the modified mISDN code are available under:
243 http://svn.digium.com/svn/thirdparty/mISDN/tags
244 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
246 libpri channel driver (chan_dahdi) DAHDI changes
247 -------------------------------------------
248 * The channel variable PRIREDIRECTREASON is now just a status variable
249 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
250 to read and alter the reason.
251 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
252 redirected-to party for an incomming redirected call if the incoming call
253 could experience further redirects. Just set the
254 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
255 to the COLR. A call has been redirected if the REDIRECTING(count) is not
257 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
258 use the inhibit(i) option on all of the REDIRECTING statements before
259 dialing the redirected-to party. You still have to set the
260 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
261 will update the redirecting-to presentation (COLR) when it becomes available.
262 * Added the ability to ignore calls that are not in a Multiple Subscriber
263 Number (MSN) list for PTMP CPE interfaces.
264 * Added dynamic range compression support for dahdi channels. It is
265 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
266 * Added support for ISDN calling and called subaddress with partial support
267 for connected line subaddress.
268 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
269 * Added handling of received HOLD/RETRIEVE messages and the optional ability
270 to transfer a held call on disconnect similar to an analog phone.
271 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
272 Will reroute/deflect an outgoing call when receive the message.
273 Can use the DAHDISendCallreroutingFacility to send the message for the
275 * Added standard location to add options to chan_dahdi dialing:
276 Dial(DAHDI/g1[/extension[/options]])
279 R Reverse charging indication
280 * Added Reverse Charging Indication (Collect calls) send/receive option.
281 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
282 Dial(DAHDI/g1/extension/R)
283 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
284 (requires latest LibPRI)
285 * Added ability to send/receive keypad digits in the SETUP message.
286 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
287 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
288 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
289 (requires latest LibPRI)
291 Asterisk Manager Interface
292 --------------------------
293 * The Hangup action now accepts a Cause header which may be used to
294 set the channel's hangup cause.
295 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
296 to specify a separate .pem file to hold a private key. By default sslcert
297 is used to hold both the public and private key.
298 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
299 for options containing the 'tls' prefix. For example, 'sslenable' is now
300 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
301 across all .conf files. All affected sample.conf files have been modified to
302 reflect this change. Previous options such as 'sslenable' still work,
303 but options with the 'tls' prefix are preferred.
304 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
305 in a channel. (res_mutestream.so)
306 * The configuration file manager.conf now supports a channelvars option, which
307 specifies a list of channel variables to include in each channel-oriented
309 * The redirect command now has new parameters ExtraContext, ExtraExtension,
310 and ExtraPriority to allow redirecting the second channel to a different
311 location than the first.
313 Channel Event Logging
314 ---------------------
315 * A new interface, CEL, is introduced here. CEL logs single events, much like
316 the AMI, but it differs from the AMI in that it logs to db backends much
317 like CDR does; is based on the event subsystem introduced by Russell, and
318 can share in all its benefits; allows multiple backends to operate like CDR;
319 is specialized to event data that would be of concern to billing sytems,
320 like CDR. Backends for logging and accounting calls have been produced,
321 but a new CDR backend is still in development.
325 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR officianados.
326 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
327 etc are performed. Thus the peices of CDR can be grouped into multilegged sets.
328 * Multiple files and formats can now be specified in cdr_custom.conf.
329 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
330 See configs/cdr_syslog.conf.sample for more information.
331 * A 'sequence' field has been added to CDRs which can be combined with
332 linkedid or uniqueid to uniquely identify a CDR.
334 Calendaring for Asterisk
335 ------------------------
336 * A new set of modules were added supporing calendar integration with Asterisk.
337 Dialplan functions for reading from and writing to calendars are included,
338 as well as the ability to execute dialplan logic upon calendar event notifications.
339 iCalendar, CalDAV, and Exchange Server calendars are supported (Exchange support
340 only tested on Exchange Server 2003 with no support for forms-based authentication).
342 Multicast RTP Support
343 ---------------------
344 * A new RTP engine and channel driver have been added which supports Multicast RTP.
345 The channel driver can be used with the Page application to perform multicast RTP
346 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
347 Type can be either basic or linksys.
348 Destination is the IP address and port for the RTP packets.
349 Control address is specific to the linksys type and is used for sending the control
350 packets unique to them.
352 Security Events Framework
353 -------------------------
354 * Asterisk has a new C API for reporting security events. The module res_security_log
355 sends these events to the "security" logger level. Currently, AMI is the only
356 Asterisk component that reports security events. However, SIP support will be
357 coming soon. For more information on the security events framework, see the
358 "Security Events" chapter of the included documentation - doc/tex/asterisk.pdf.
362 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
363 XMPP text messages to the remote JID.
364 * Modules.conf has a new option - "require" - that marks a module as critical for
365 the execution of Asterisk.
366 If one of the required modules fail to load, Asterisk will exit with a return
368 * An 'X' option has been added to the asterisk application which enables #exec support.
369 This allows #exec to be used in asterisk.conf.
370 * jabber.conf supports a new option auth_policy that toggles auto user registration.
372 ------------------------------------------------------------------------------
373 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
374 ------------------------------------------------------------------------------
378 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
379 Snom phones use this for call pickup of extensions that the phone is
381 * Added support for setting the domain in the URI for caller of an
382 outbound call by using the SIPFROMDOMAIN channel variable.
383 * Added a new configuration option "remotesecret" for authentication to
384 remote services. For backwards compatibility, "secret" still has the
385 same function as before, but now you can configure both a remote secret and a
386 local secret for mutual authentication.
387 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
388 the sound will be played to the target of an attended transfer
389 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
390 finer control over how many peers Asterisk will qualify and the gap between them
391 when all peers need to be qualified at the same time.
392 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
393 (either globally or for a specific peer), chan_sip will treat any SDP data
394 it receives as new data and update the media stream accordingly. By
395 default, Asterisk will only modify the media stream if the SDP session
396 version received is different from the current SDP session version. This
397 option is required to interoperate with devices that have non-standard SDP
398 session version implementations (observed with Microsoft OCS). This option
399 is disabled by default.
400 * The parsing of register => lines in sip.conf has been modified to allow a port
401 to be present in the "user" portion. Please see the sip.conf.sample file for more
403 * Added support for subscribing to MWI on a remote server and making the status available
404 as a mailbox. Please see the sip.conf.sample file for more information.
405 * Added a function to remove SIP headers added in the dialplan before the
406 first INVITE is generated - SIPRemoveHeader()
407 * Channel variables set with setvar= in a device configuration is now
408 set both for inbound and outbound calls.
409 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
413 * Added immediate option to iax.conf
414 * Added forceencryption option to iax.conf
415 * Added Encryption and Trunk status to manager command "iaxpeers"
419 * The configuration file now holds separate sections for devices and lines.
420 Please have a look at configs/skinny.conf.sample and change your skinny.conf
425 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
426 support for LibOpenR2. http://www.libopenr2.org/
427 * The UK option waitfordialtone has been added for use with BT analog
429 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
430 is used in conjunction with the 'faxdetect' configuration option. When
431 'faxbuffers' is used and fax tones are detected, the channel will dynamically
432 switch to the configured faxbuffers policy. For example, to use 6 buffers
433 and a 'full' buffer policy for a fax transmission, add:
435 The faxbuffers configuration will be in affect until the call is torn down.
436 * Added service message support for 4ESS/5ESS switches.
440 * Added a new dialplan function, CURLOPT, which permits setting various
441 options that may be useful with the CURL dialplan function, such as
442 cookies, proxies, connection timeouts, passwords, etc.
443 * Permit the syntax and synopsis fields of the corresponding dialplan
444 functions to be individually set from func_odbc.conf.
445 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
446 * func_odbc now may specify an insert query to execute, when the write query
447 affects 0 rows (usually indicating that no such row exists).
448 * Added a new dialplan function, LISTFILTER, which permits removing elements
449 from a set list, by name. Uses the same general syntax as the existing CUT
450 and FIELDQTY dialplan functions, which also manage lists.
451 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
452 obtaining realtime data from the dialplan.
453 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
454 a subroutine when using the GoSub() and Return() applications.
455 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
456 of "core show function AUDIOHOOK_INHERIT" from the CLI
457 * Added AES_ENCRYPT. For information on its use, please see the output
458 of "core show function AES_ENCRYPT" from the CLI
459 * Added AES_DECRYPT. For information on its use, please see the output
460 of "core show function AES_DECRYPT" from the CLI
461 * func_odbc now supports database transactions across multiple queries.
465 * Scheduled meetme conferences may now have their end times extended by
467 * app_authenticate now gives the ability to select a prompt other than
469 * app_directory now pays attention to the searchcontexts setting in
470 voicemail.conf and will look through all contexts, if no context is
471 specified in the initial argument.
472 * A new application, Originate, has been introduced, that allows asynchronous
473 call origination from the dialplan.
474 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
475 in addition to the setting in the "general" context.
476 * Added ConfBridge dialplan application which does conference bridges without
477 DAHDI. For information on its use, please see the output of
478 "core show application ConfBridge" from the CLI.
482 * The Asterisk CLI has a new command, "channel redirect", which is similar in
483 operation to the AMI Redirect action.
484 * extensions.conf now allows you to use keyword "same" to define an extension
485 without actually specifying an extension. It uses exactly the same pattern
486 as previously used on the last "exten" line. For example:
487 exten => 123,1,NoOp(something)
488 same => n,SomethingElse()
489 * musiconhold.conf classes of type 'files' can now use relative directory paths,
490 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
491 * All deprecated CLI commands are removed from the sourcecode. They are now handled
492 by the new clialiases module. See cli_aliases.conf.sample file.
493 * Times within timespecs are now accurate down to the minute. This is a change
494 from historical Asterisk, which only provided timespecs rounded to the nearest
495 even (read: evenly divisible by 2) minute mark.
496 * The realtime switch now supports an option flag, 'p', which disables searches for
498 * In addition to a time range and date range, timespecs now accept a 5th optional
499 argument, timezone. This allows you to perform time checks on alternate
500 timezones, especially if those daylight savings time ranges vary from your
501 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
503 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
504 give you the correct output for an asterisk box behind nat. It will give you the
505 externhost and localnet settings.
506 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
507 can connect calls in passthrough mode, as well as record and play back files.
508 * Successful and unsuccessful call pickup can now be alerted through sounds, by
509 using pickupsound and pickupfailsound in features.conf.
510 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
511 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
512 instead of the /var/run/asterisk.pid where it used to be. This will make
513 installs as non-root easier to manage.
515 Asterisk Manager Interface
516 --------------------------
517 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
518 a non-empty value) in your request. If you do this, any pending AMI events will
519 *not* be included in the response to your request as they would normally, but
520 will be left in the event queue for the next request you make to retrieve. For
521 some applications, this will allow you to guarantee that you will only see
522 events in responses to 'WaitEvent' actions, and can better know when to expect them.
523 To know whether the Asterisk server supports this header or not, your client can
524 inspect the first response back from the server to see if it includes this header:
526 Pragma: SuppressEvents
528 If this is included, the server supports event suppression.
530 * Added 4 new Actions to list skinny device(s) and line(s)
536 LDAP Schema File Additions
537 --------------------------
538 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
539 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
541 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
542 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
543 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
544 * Removed redundant IPaddr (there's already IPAddress)
545 - Gives more configuration Flags for SIP-Users available (tested)
546 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
547 without extensibleObject (which really should be the last resort); gives
548 also additional possibilities for LDAP-filter
550 ------------------------------------------------------------------------------
551 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
552 ------------------------------------------------------------------------------
554 Device State Handling
555 ---------------------
556 * The event infrastructure in Asterisk got another big update to help support
557 distributed events. It currently supports distributed device state and
558 distributed Voicemail MWI (Message Waiting Indication). A new module has
559 been merged, res_ais, which facilitates communicating events between servers.
560 It uses the SAForum AIS (Service Availability Forum Application Interface
561 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
562 a cluster of Asterisk servers, and to share events between them. For more
563 information on setting this up, see doc/distributed_devstate.txt.
567 * Added a new dialplan function, AST_CONFIG(), which allows you to access
568 variables from an Asterisk configuration file.
569 * The JACK_HOOK function now has a c() option to supply a custom client name.
570 * Added two new dialplan functions from libspeex for audio gain control and
571 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
572 rx directions of a channel from the dialplan.
573 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
574 based on other parameters. The default is still to search based on the
575 forwarding station ID. However, there are new options that allow you to search
576 based on the message desk terminal ID, or the message desk number.
577 * TIMEOUT() has been modified to be accurate down to the millisecond.
578 * ENUM*() functions now include the following new options:
579 - 'u' returns the full URI and does not strip off the URI-scheme.
580 - 's' triggers ISN specific rewriting
581 - 'i' looks for branches into an Infrastructure ENUM tree
582 - 'd' for a direct DNS lookup without any flipping of digits.
583 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
584 * CHANNEL() now has options for the maximum, minimum, and standard or normal
585 deviation of jitter, rtt, and loss for a call using chan_sip.
587 DAHDI channel driver (chan_dahdi) Changes
588 ----------------------------------------
589 * Channels can now be configured using named sections in chan_dahdi.conf, just
590 like other channel drivers, including the use of templates.
591 * The default for pridialplan has changed from 'national' to 'unknown'.
595 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
596 to something that matches the pattern a hint will be created using the contents
597 and variables evaluated.
598 * Dialplan matching has been extended to allow an extension to return to the
599 PBX core to wait for more digits. This is done by using the new dialplan
600 application called "Incomplete". This will permit a whole new level of
601 extension control, by giving the administrator more control over early
602 matches employing one of the short-circuit pattern match operators. Note
603 that custom applications can trigger this same behavior by returning the
604 special value AST_PBX_INCOMPLETE.
608 * Directory now permits both first and last names to be matched at the same
609 time. In addition, the number of digits to enter of the name can be set in
610 the arguments to Directory; previously, you could enter only 3, regardless
611 of how many names are in your company. For large companies, this should be
613 * Voicemail now permits a mailbox setting to wrap around from first to last
614 messages, if the "messagewrap" option is set to a true value.
615 * Voicemail now permits an external script to be run, for password validation.
616 The script should output "VALID" or "INVALID" on stdout, depending upon the
617 wish to validate or invalidate the password given. Arguments are:
618 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
620 * Dial has a new option: F(context^extension^pri), which permits a callee to
621 continue in the dialplan, at the specified label, if the caller hangs up.
622 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
623 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
624 * The Jack application now has a c() option to supply a custom client name.
625 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
626 like the pre-existing whisper mode, except that the spy can also talk to the
627 participant on the bridged channel as well.
628 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
629 to be spoken instead of the channel name or number. For more information on the
630 use of this option, issue the command "core show application ChanSpy" from the
632 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
633 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
634 words, if using the 'd' option, it is not possible to enter a number to append to
635 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
636 change to whisper mode, and pressing 6 will change to barge mode.
637 * ExternalIVR now takes several options that affect the way it performs, as
638 well as having several new commands. Please see doc/externalivr.txt for the
639 complete documentation.
640 * Added ability to communicate over a TCP socket instead of forking a child process for the
641 ExternalIVR application.
642 * ChanIsAvail has a new option, 'a', which will return all available channels instead
643 of just the first one if you give the function more then one channel to check.
644 * PrivacyManager now takes an option where you can specify a context where the
645 given number will be matched. This way you have more control over who is allowed
646 and it stops the people who blindly enter 10 digits.
647 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
648 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
649 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
650 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
651 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
652 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
653 * The Dial() application no longer copies the language used by the caller to the callee's
654 channel. If you desire for the caller's channel's language to be used for file playback
655 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
656 * SendImage() no longer hangs up the channel on error; instead, it sets the
657 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
658 'UNSUPPORTED'. This change makes SendImage() more consistent with other
660 * Park has a new option, 's', which silences the announcement of the parking space number.
661 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
662 invalid input and will be assumed to mean that no timeout is desired.
666 * Added DNS manager support to registrations for peers referencing peer entries.
667 DNS manager runs in the background which allows DNS lookups to be run asynchronously
668 as well as periodically updating the IP address. These properties allow for
669 better performance as well as recovery in the event of an IP change.
670 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
671 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
672 These changes also provide performance improvements for call setup and tear down.
673 * Added ability to specify registration expiry time on a per registration basis in
675 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
677 * Added t38pt_usertpsource option. See sip.conf.sample for details.
678 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
679 * 'sip show peers' and 'sip show users' display their entries sorted in
680 alphabetical order, as opposed to the order they were in, in the config
682 * Videosupport now supports an additional option, "always", which always sets
683 up video RTP ports, even on clients that don't support it. This helps with
684 callfiles and certain transfers to ensure that if two video phones are
685 connected, they will always share video feeds.
689 * Existing DNS manager lookups extended to check for SRV records.
690 * IAX2 encryption support has been improved to support periodic key rotation
691 within a call for enhanced security. The option "keyrotate" has been
692 provided to disable this functionality to preserve backwards compatibility
693 with older versions of IAX2 that do not support key rotation.
697 * New CLI command, "config reload <file.conf>" which reloads any module that
698 references that particular configuration file. Also added "config list"
699 which shows which configuration files are in use.
700 * New CLI commands, "pri show version" and "ss7 show version" that will
701 display which version of libpri and libss7 are being used, respectively.
702 A new API call was added so trunk will now have to be compiled against
703 a versions of libpri and libss7 that have them or it will not know that
704 these libraries exist.
705 * The commands "core show globals", "core set global" and "core set chanvar" has
706 been deprecated in favor of the more semanticly correct "dialplan show globals",
707 "dialplan set chanvar" and "dialplan set global".
708 * New CLI command "dialplan show chanvar" to list all variables associated
709 with a given channel.
713 * Addresses managed by DNS manager now can check to see if there is a DNS
714 SRV record for a given domain and will use that hostname/port if present.
716 AMI - The manager (TCP/TLS/HTTP)
717 --------------------------------
718 * The Status command now takes an optional list of variables to display
719 along with channel status.
720 * The QueueEntry event now also includes the channel's uniqueid
724 * res_odbc no longer has a limit of 1023 total possible unshared connections,
725 as some people were running into this limit. This limit has been increased
730 * The TRANSFER queue log entry now includes the the caller's original
731 position in the transferred-from queue.
732 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
733 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
734 as well as an explanation about timeout options in general
735 * Added a new option - C - for forcing the "answered elsewhere" flag on
736 cancellation of calls in to members of the queue. This is to avoid the
737 call to a member of a queue having the call listed as a "missed call".
741 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
742 adaptive capabilities. What this means in practical terms is that if your
743 realtime table lacks critical fields, Asterisk will now emit warnings to
744 that effect. Also, some of the realtime drivers have the ability (if
745 configured) to automatically add those columns to the table with the
746 correct type and length.
750 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
751 the 'setvar' option to cause a given audio file to be played upon completion
752 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
753 Skinny channels only.
754 * You can now compile Asterisk against the Hoard Memory Allocator, see doc/hoard.txt
755 for more information.
756 * Config file variables may now be appended to, by using the '+=' append
757 operator. This is most helpful when working with long SQL queries in
758 func_odbc.conf, as the queries no longer need to be specified on a single
760 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
761 which will add a second to the billsec when the ending
762 time is set, if the number in the microseconds field of the end time is
763 greater than the number of microseconds in the answer time. This allows
764 users to count the 'initiated' seconds in their billing records.
766 ------------------------------------------------------------------------------
767 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
768 ------------------------------------------------------------------------------
770 AMI - The manager (TCP/TLS/HTTP)
771 --------------------------------
772 * Manager has undergone a lot of changes, all of them documented
773 in doc/manager_1_1.txt
774 * Manager version has changed to 1.1
775 * Added a new action 'CoreShowChannels' to list currently defined channels
776 and some information about them.
777 * Added a new action 'SIPshowregistry' to list SIP registrations.
778 * Added TLS support for the manager interface and HTTP server
779 * Added the URI redirect option for the built-in HTTP server
780 * The output of CallerID in Manager events is now more consistent.
781 CallerIDNum is used for number and CallerIDName for name.
782 * Enable https support for builtin web server.
783 See configs/http.conf.sample for details.
784 * Added a new action, GetConfigJSON, which can return the contents of an
785 Asterisk configuration file in JSON format. This is intended to help
786 improve the performance of AJAX applications using the manager interface
788 * SIP and IAX manager events now use "ChannelType" in all cases where we
789 indicate channel driver. Previously, we used a mixture of "Channel"
790 and "ChannelDriver" headers.
791 * Added a "Bridge" action which allows you to bridge any two channels that
792 are currently active on the system.
793 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
794 the voicemail users setup.
795 * Added 'DBDel' and 'DBDelTree' manager commands.
796 * cdr_manager now reports events via the "cdr" level, separating it from
797 the very verbose "call" level.
798 * Manager users are now stored in memory. If you change the manager account
799 list (delete or add accounts) you need to reload manager.
800 * Added Masquerade manager event for when a masquerade happens between
802 * Added "manager reload" command for the CLI
803 * Lots of commands that only provided information are now allowed under the
804 Reporting privilege, instead of only under Call or System.
805 * The IAX* commands now require either System or Reporting privilege, to
806 mirror the privileges of the SIP* commands.
807 * Added ability to retrieve list of categories in a config file.
808 * Added ability to retrieve the content of a particular category.
809 * Added ability to empty a context.
810 * Created new action to create a new file.
811 * Updated delete action to allow deletion by line number with respect to category.
812 * Added new action insert to add new variable to category at specified line.
813 * Updated action newcat to allow new category to be inserted in file above another
815 * Added new event "JitterBufStats" in the IAX2 channel
816 * Originate now requires the Originate privilege and, if you want to call out
817 to a subshell, it requires the System privilege, as well. This was done to
818 enhance manager security.
819 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
820 * New command: Atxfer. See doc/manager_1_1.txt for more details or
821 manager show command Atxfer from the CLI
822 * New command: IAXregistry. See doc/manager_1_1.txt for more details or
823 manager show command IAXregistry from the CLI
827 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
828 state in the dialplan, as well as creating custom device states that are
829 controllable from the dialplan.
830 * Extend CALLERID() function with "pres" and "ton" parameters to
831 fetch string representation of calling number presentation indicator
832 and numeric representation of type of calling number value.
833 * MailboxExists converted to dialplan function
834 * A new option to Dial() for telling IP phones not to count the call
835 as "missed" when dial times out and cancels.
836 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
837 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
838 held for any given channel. Also, locks are automatically freed when a
840 * Added HINT() dialplan function that allows retrieving hint information.
841 Hints are mappings between extensions and devices for the sake of
842 determining the state of an extension. This function can retrieve the list
843 of devices or the name associated with a hint.
844 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
846 * Added SYSINFO() dialplan function which allows retrieval of system information
847 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
848 the existence of a dialplan target.
849 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
850 upper and lower case, respectively.
851 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
852 ID for the call (not the Asterisk call ID or unique ID), provided that the
853 channel driver supports this. For SIP, you get the SIP call-ID for the
854 bridged channel which you can store in the CDR with a custom field.
858 * Added CLI permissions, config file: cli_permissions.conf
859 default is to allow all commands for every local user/group.
860 Also this new feature added three new CLI commands:
861 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
862 - cli reload permissions
863 - cli show permissions
864 * New CLI command "core show hint" (usage: core show hint <exten>)
865 * New CLI command "core show settings"
866 * Added 'core show channels count' CLI command.
867 * Added the ability to set the core debug and verbose values on a per-file basis.
868 * Added 'queue pause member' and 'queue unpause member' CLI commands
869 * Ability to set process limits ("ulimit") without restarting Asterisk
870 * Enhanced "agi debug" to print the channel name as a prefix to the debug
871 output to make debugging on busy systems much easier.
872 * New CLI commands "dialplan set extenpatternmatching true/false"
873 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
874 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
875 listed in the startup_commands section of cli.conf will get executed.
876 * Added a CLI command, "devstate change", which allows you to set custom device
877 states from the func_devstate module that provides the DEVICE_STATE() function
878 and handling of the "Custom:" devices.
879 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
880 sorted into the different possible callbacks, with the number of entries
881 currently scheduled for each. Gives you a feel for how busy the sip channel
883 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
884 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
885 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
889 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
890 option is enabled, Asterisk will watch for a CNG tone in the incoming audio
891 for a received call. If it is detected, the channel will jump to the
892 'fax' extension in the dialplan.
893 * Improved NAT and STUN support.
894 chan_sip now can use port numbers in bindaddr, externip and externhost
895 options, as well as contact a STUN server to detect its external address
896 for the SIP socket. See sip.conf.sample, 'NAT' section.
897 * The default SIP useragent= identifier now includes the Asterisk version
898 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
899 If set, and the incoming request carries authentication info,
900 the username to match in the users list is taken from the Digest header
901 rather than from the From: field. This feature is considered experimental.
902 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
903 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
904 * The "localmask" setting was removed in version 1.2 and the reminder about it
905 being removed is now also removed.
906 * A new option "busylevel" for setting a level of calls where asterisk reports
907 a device as busy, to separate it from call-limit. This value is also added
908 to the SIP_PEER dialplan function.
909 * A new realtime family called "sipregs" is now supported to store SIP registration
910 data. If this family is defined, "sippeers" will be used for configuration and
911 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
912 registration data, as before.
913 * The SIPPEER function have new options for port address, call and pickup groups
914 * Added support for T.140 realtime text in SIP/RTP
915 * The "checkmwi" option has been removed from sip.conf, as it is no longer
916 required due to the restructuring of how MWI is handled. See the descriptions
917 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
918 for more information.
919 * Added rtpdest option to CHANNEL() dialplan function.
920 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
921 * SIP now adds a header to the CANCEL if the call was answered by another phone
922 in the same dial command, or if the new c option in dial() is used.
923 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
924 states it is not needed. For phones, however, that do require it the "registertrying" option
925 has been added so it can be enabled.
926 * A new option called "callcounter" (global/peer/user level) enables call counters needed
927 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
928 used to enable this functionality).
929 * New settings for timer T1 and timer B on a global level or per device. This makes it
930 possible to force timeout faster on non-responsive SIP servers. These settings are
931 considered advanced, so don't use them unless you have a problem.
932 * Added a dial string option to be able to set the To: header in an INVITE to any
934 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
935 the qualify frequency.
936 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
937 were not properly torn down due to network or endpoint failures during an established
939 * Added experimental TCP and TLS support for SIP. See doc/siptls.txt and
940 configs/sip.conf.sample for more information on how it is used.
941 * Added a new configuration option "authfailureevents" that enables manager events when
942 a peer can't authenticate properly.
943 * Added DNS manager support to registrations for peers not referencing a peer entry.
947 * Added the trunkmaxsize configuration option to chan_iax2.
948 * Added the srvlookup option to iax.conf
949 * Added support for OSP. The token is set and retrieved through the CHANNEL()
952 XMPP Google Talk/Jingle changes
953 -------------------------------
954 * Added the bindaddr option to gtalk.conf.
958 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
959 * Proper codec support in chan_skinny.
960 * Added settings for IP and Ethernet QoS requests
964 * Added separate settings for media QoS in mgcp.conf
966 Console Channel Driver changes
967 ------------------------------
968 * Added experimental support for video send & receive to chan_oss.
969 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
972 Phone channel changes (chan_phone)
973 ----------------------------------
974 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
976 H.323 channel Changes
977 ---------------------
978 * H323 remote hold notification support added (by NOTIFY message
979 and/or H.450 supplementary service)
981 Local channel changes
982 ---------------------
983 * The device state functionality in the Local channel driver has been updated
984 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
985 to just UNKNOWN if the extension exists.
986 * Added jitterbuffer support for chan_local. This allows you to use the
987 generic jitterbuffer on incoming calls going to Asterisk applications.
988 For example, this would allow you to use a jitterbuffer for an incoming
989 SIP call to Voicemail by putting a Local channel in the middle. This
990 feature is enabled by using the 'j' option in the Dial string to the Local
991 channel in conjunction with the existing 'n' option for local channels.
992 * A 'b' option has been added which causes chan_local to return the actual channel
993 that is behind it when queried. This is useful for transfer scenarios as the
994 actual channel will be transferred, not the Local channel.
996 Agent channel changes
997 ----------------------
998 * The ackcall and endcall options are now supplemented with options acceptdtmf
999 and enddtmf. These allow for the DTMF keypress to be configurable. The options
1000 default to their old hard-coded values ('#' and '*' respectively) so this should
1001 not break any existing agent installations.
1003 DAHDI channel driver (chan_dahdi) Changes
1004 ----------------------------------------
1005 * SS7 support (via libss7 library)
1006 * In India, some carriers transmit CID via dtmf. Some code has been added
1007 that will handle some situations. The cidstart=polarity_IN choice has been added for
1008 those carriers that transmit CID via dtmf after a polarity change.
1009 * CID matching information is now shown when doing 'dialplan show'.
1010 * Added dahdi show version CLI command.
1011 * Added setvar support to chan_dahdi.conf channel entries.
1012 * Added two new options: mwimonitor and mwimonitornotify. These options allow
1013 you to enable MWI monitoring on FXO lines. When the MWI state changes,
1014 the script specified in the mwimonitornotify option is executed. An internal
1015 event indicating the new state of the mailbox is also generated, so that
1016 the normal MWI facilities in Asterisk work as usual.
1017 * Added signalling type 'auto', which attempts to use the same signalling type
1018 for a channel as configured in DAHDI. This is primarily designed for analog
1019 ports, but will also work for digital ports that are configured for FXS or FXO
1020 signalling types. This mode is also the default now, so if your chan_dahdi.conf
1021 does not specify signalling for a channel (which is unlikely as the sample
1022 configuration file has always recommended specifying it for every channel) then
1023 the 'auto' mode will be used for that channel if possible.
1024 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
1025 state for a channel; also ensured that the DNDState Manager event is
1026 emitted no matter how the DND state is set or cleared.
1030 * Added a new channel driver, chan_unistim. See doc/unistim.txt and
1031 configs/unistim.conf.sample for details. This new channel driver allows
1032 you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
1033 * Added a new channel driver, chan_console, which uses portaudio as a cross
1034 platform audio interface. It was written as a channel driver that would
1035 work with Mac CoreAudio, but portaudio supports a number of other audio
1036 interfaces, as well. Note that this channel driver requires v19 or higher
1037 of portaudio; older versions have a different API.
1041 * Added the ability to specify arguments to the Dial application when using
1042 the DUNDi switch in the dialplan.
1043 * Added the ability to set weights for responses dynamically. This can be
1044 done using a global variable or a dialplan function. Using the SHELL()
1045 function would allow you to have an external script set the weight for
1047 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
1048 functions will allow you to initiate a DUNDi query from the dialplan,
1049 find out how many results there are, and access each one.
1053 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
1054 functions will allow you to initiate an ENUM lookup from the dialplan,
1055 and Asterisk will cache the results. ENUMRESULT can be used to access
1056 the results without doing multiple DNS queries.
1060 * Added the ability to customize which sound files are used for some of the
1061 prompts within the Voicemail application by changing them in voicemail.conf
1062 * Added the ability for the "voicemail show users" CLI command to show users
1063 configured by the dynamic realtime configuration method.
1064 * MWI (Message Waiting Indication) handling has been significantly
1065 restructured internally to Asterisk. It is now totally event based
1066 instead of polling based. The voicemail application will notify other
1067 modules that have subscribed to MWI events when something in the mailbox
1069 This also means that if any other entity outside of Asterisk is changing
1070 the contents of mailboxes, then the voicemail application still needs to
1071 poll for changes. Examples of situations that would require this option
1072 are web interfaces to voicemail or an email client in the case of using
1073 IMAP storage. So, two new options have been added to voicemail.conf
1074 to account for this: "pollmailboxes" and "pollfreq". See the sample
1075 configuration file for details.
1076 * Added "tw" language support
1077 * Added support for storage of greetings using an IMAP server
1078 * Added ability to customize forward, reverse, stop, and pause keys for message playback
1079 * SMDI is now enabled in voicemail using the smdienable option.
1080 * A "lockmode" option has been added to asterisk.conf to configure the file
1081 locking method used for voicemail, and potentially other things in the
1082 future. The default is the old behavior, lockfile. However, there is a
1083 new method, "flock", that uses a different method for situations where the
1084 lockfile will not work, such as on SMB/CIFS mounts.
1085 * Added the ability to backup deleted messages, to ease recovery in the case
1086 that a user accidentally deletes a message, and discovers that they need it.
1087 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
1088 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
1089 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
1090 voicemail boxes. The SMDI interface can also poll for MWI changes when some
1091 outside entity is modifying the state of the mailbox (such as IMAP storage or
1092 a web interface of some kind).
1093 * Added the support for marking messages as "urgent." There are two methods to accomplish
1094 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
1095 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
1096 the message as urgent after he has recorded a voicemail by following the voice instructions.
1097 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
1102 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
1103 used across multiple queues.
1104 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
1105 setqueueentryvar options for each queue, see queues.conf.sample for details.
1106 * Added keepstats option to queues.conf which will keep queue
1107 statistics during a reload.
1108 * setinterfacevar option in queues.conf also now sets a variable
1109 called MEMBERNAME which contains the member's name.
1110 * Added 'Strategy' field to manager event QueueParams which represents
1111 the queue strategy in use.
1112 * Added option to run macro when a queue member is connected to a caller,
1113 see queues.conf.sample for details.
1114 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
1115 does not count paused queue members as unavailable.
1116 * Added min-announce-frequency option to queues.conf which allows you to control the
1117 minimum amount of time between queue announcements for use when the caller's queue
1118 position changes frequently.
1119 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
1121 * Added ability for non-realtime queues to have realtime members
1122 * Added the "linear" strategy to queues.
1123 * Added the "wrandom" strategy to queues.
1124 * Added new channel variable QUEUE_MIN_PENALTY
1125 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
1126 rules in queuerules.conf. See configs/queuerules.conf.sample for details
1127 * Added a new parameter for member definition, called state_interface. This may be
1128 used so that a member may be called via one interface but have a different interface's
1129 device state reported.
1130 * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
1131 "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
1132 "manager show command QueueReset."
1133 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
1134 specified by the periodic-announce option, then one will be chosen randomly when it is time
1135 to play a periodic announcment
1136 * New configuration options: announce-position now takes two more values in addition to "yes" and
1137 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
1138 announce-position-limit. By setting announce-position to "limit" callers will only have their
1139 position announced if their position is less than what is specified by announce-position-limit.
1140 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
1141 will be told that their are more than announce-position-limit callers waiting.
1142 * Two new queue log events have been added. An ADDMEMBER event will be logged
1143 when a realtime queue member is added and a REMOVEMEMBER event will be logged
1144 when a realtime queue member is removed. Since there is no calling channel associated
1145 with these events, the string "REALTIME" is placed where the channel's unique id
1146 is typically placed.
1147 * The configuration method for the "joinempty" and "leavewhenempty" options has
1148 changed to a comma-separated list of methods of determining member availability
1149 instead of vague terms such as "yes," "loose," "no," and "strict." These old four
1150 values are still accepted for backwards-compatibility, though.
1151 * The average talktime is now calculated on queues. This information is reported via the
1152 CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
1153 and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
1158 * The 'o' option to provide an optimization has been removed and its functionality
1159 has been enabled by default.
1160 * When a conference is created, the UNIQUEID of the channel that caused it to be
1161 created is stored. Then, every channel that joins the conference will have the
1162 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
1163 callers that come and go from long standing conferences.
1164 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
1165 except it does operations on a channel by name, instead of number in a conference.
1166 This is a very useful feature in combination with the 'X' option to ChanSpy.
1167 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
1169 * Added new RealTime functionality to provide support for scheduled conferencing.
1170 This includes optional messages to the caller if they attempt to join before
1171 the schedule start time, or to allow the caller to join the conference early.
1172 Also included is optional support for limiting the number of callers per
1173 RealTime conference.
1174 * Added the S() and L() options to the MeetMe application. These are pretty
1175 much identical to the S() and L() options to Dial(). They let you set
1176 timeouts for the conference, as well as have warning sounds played to
1177 let the caller know how much time is left, and when it is running out.
1178 * Added the ability to do "meetme concise" with the "meetme" CLI command.
1179 This extends the concise capabilities of this CLI command to include
1180 listing all conferences, instead of an addition to the other sub commands
1181 for the "meetme" command.
1182 * Added the ability to specify the music on hold class used to play into the
1183 conference when there is only one member and the M option is used.
1184 * Added MEETME_INFO dialplan function which provides a way to query
1185 various properties of a Meetme conference.
1187 Other Dialplan Application Changes
1188 ----------------------------------
1189 * Argument support for Gosub application
1190 * From the to-do lists: straighten out the app timeout args:
1191 Wait() app now really does 0.3 seconds- was truncating arg to an int.
1192 WaitExten() same as Wait().
1193 Congestion() - Now takes floating pt. argument.
1194 Busy() - now takes floating pt. argument.
1195 Read() - timeout now can be floating pt.
1196 WaitForRing() now takes floating pt timeout arg.
1197 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
1198 * Added 's' option to Page application.
1199 * Added an optional timeout argument to the Page application.
1200 * Added 'E', 'V', and 'P' commands to ExternalIVR.
1201 * Added 'o' and 'X' options to Chanspy.
1202 * Added a new dialplan application, Bridge, which allows you to bridge the
1203 calling channel to any other active channel on the system.
1204 * Added the ability to specify a music on hold class to play instead of ringing
1205 for the SLATrunk application.
1206 * The Read application no longer exits the dialplan on error. Instead, it sets
1207 READSTATUS to ERROR, which you can catch and handle separately.
1208 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
1209 of asking for verification of each name, one at a time.
1210 * Privacy() no longer uses privacy.conf, as all options are specifyable as
1211 direct options to the app.
1212 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
1214 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
1215 * The ChannelRedirect application no longer exits the dialplan if the given channel
1216 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
1217 or NOCHANNEL if the given channel was not found.
1218 * The silencethreshold setting that was previously configurable in multiple
1219 applications is now settable globally via dsp.conf.
1221 Music On Hold Changes
1222 ---------------------
1223 * A new option, "digit", has been added for music on hold classes in
1224 musiconhold.conf. If this is set for a music on hold class, a caller
1225 listening to music on hold can press this digit to switch to listening
1226 to this music on hold class.
1227 * Support for realtime music on hold has been added.
1228 * In conjunction with the realtime music on hold, a general section has
1229 been added to musiconhold.conf, its sole variable is cachertclasses. If this
1230 is set, then music on hold classes found in realtime will be cached in memory.
1234 * AEL upgraded to use the Gosub with Arguments instead
1235 of Macro application, to hopefully reduce the problems
1236 seen with the artificially low stack ceiling that
1237 Macro bumps into. Macros can only call other Macros
1238 to a depth of 7. Tests run using gosub, show depths
1239 limited only by virtual memory. A small test demonstrated
1240 recursive call depths of 100,000 without problems.
1241 -- in addition to this, all apps that allowed a macro
1242 to be called, as in Dial, queues, etc, are now allowing
1243 a gosub call in similar fashion.
1244 * AEL now generates LOCAL(argname) declarations when it
1245 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
1246 etc. That makes the arguments local in scope. The user
1247 can define their own local variables in macros, now,
1248 by saying "local myvar=someval;" or using Set() in this
1249 fashion: Set(LOCAL(myvar)=someval); ("local" is now
1251 * utils/conf2ael introduced. Will convert an extensions.conf
1252 file into extensions.ael. Very crude and unfinished, but
1253 will be improved as time goes by. Should be useful for a
1254 first pass at conversion.
1255 * aelparse will now read extensions.conf to see if a referenced
1256 macro or context is there before issueing a warning.
1257 * AEL parser sets a local channel variable ~~EXTEN~~, to
1258 preserve the value of ${EXTEN} thru switch statements.
1259 * New operator in $[...] expressions: the ~~ operator serves
1260 as a concatenation operator. AT THE MOMENT, it is really only
1261 necessary and useful in AEL, especially in if() expressions.
1262 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
1263 any enclosing double-quotes, and evaluate to the value of a
1264 concatenated with the value of b. For example if a is set to
1265 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
1266 evaluate to xyzabc .
1269 Call Features (res_features) Changes
1270 ------------------------------------
1271 * Added the parkedcalltransfers option to features.conf
1272 * Added parkedcallparking option to control one touch parking w/ parking
1274 * Added parkedcallhangup option to control disconnect feature w/ parking
1276 * Added parkedcallrecording option to control one-touch record w/ parking
1278 * Added BRIDGE_FEATURES variable to set available features for a channel
1279 * The built-in method for doing attended transfers has been updated to
1280 include some new options that allow you to have the transferee sent
1281 back to the person that did the transfer if the transfer is not successful.
1282 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
1283 in features.conf.sample.
1284 * Added support for configuring named groups of custom call features in
1285 features.conf. This means that features can be written a single time, and
1286 then mapped into groups of features for different key mappings or easier
1288 * Updated the ParkedCall application to allow you to not specify a parking
1289 extension. If you don't specify a parking space to pick up, it will grab
1290 the first one available.
1291 * Added cli command 'features reload' to reload call features from features.conf
1292 * Moved into core asterisk binary.
1293 * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
1295 Language Support Changes
1296 ------------------------
1297 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
1298 * Added support for the Hungarian language for saying numbers, dates, and times.
1302 * Added SPEECH commands for speech recognition. A complete listing can be found
1304 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
1305 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
1306 does not behave as expected; the native command needs to be used, instead.
1310 * Added rotatestrategy option to logger.conf, along with two new options:
1311 "timestamp" which will use the time to name the logger files instead of
1312 sequence number; and "rotate", which rotates the names of the log files,
1313 similar to the way syslog rotates files.
1314 * Added exec_after_rotate option to logger.conf, which allows a system
1315 command to be run after rotation. This is primarily useful with
1316 rotatestrategy=rotate, to allow a limit on the number of log files kept
1317 and to ensure that the oldest log file gets deleted.
1318 * Added realtime support for the queue log
1322 * The cdr_manager module has a [mappings] feature, like cdr_custom,
1323 to add fields to the manager event from the CDR variables.
1324 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
1325 backend database CDR table. Specifically, additional, non-standard
1326 columns are supported, merely by setting the corresponding CDR variable in
1327 your dialplan. In addition, you may alias any column to another name (for
1328 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
1329 simply "alias src => ANI" in the configuration file). Records may be
1330 posted to more than one backend, simply by specifying multiple categories
1331 in the configuration file. And finally, you may filter which CDRs get
1332 posted to each backend, by specifying a filter (which the record must
1333 match) for the particular category. Filters are additive (meaning all
1334 rules must match to post that CDR).
1335 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
1336 module. Specifically, you may add additional columns into the table and
1337 they will be set, if you set the corresponding CDR variable name. Also,
1338 if you omit columns in your database table, they will be silently skipped
1339 (but a record will still be inserted, based on what columns remain). Note
1340 that the other two features from cdr_adaptive_odbc (alias and filter) are
1341 not currently supported.
1342 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
1343 has been disabled using the NoCDR application.
1345 Miscellaneous New Modules
1346 -------------------------
1347 * Added a new CDR module, cdr_sqlite3_custom.
1348 * Added a new realtime configuration module, res_config_sqlite
1349 * Added a new codec translation module, codec_resample, which re-samples
1350 signed linear audio between 8 kHz and 16 kHz to help support wideband
1352 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
1353 based on configuration templates that use Asterisk dialplan function and
1354 variable substitution. It should be possible to create phone profiles and
1355 templates that work for the majority of phones provisioned over http. It
1356 is currently only intended to provision a single user account per phone.
1357 An example profile and set of templates for Polycom phones is provided.
1358 NOTE: Polycom firmware is not included, but should be placed in
1359 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
1360 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
1361 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
1362 provided; there is a JACK() application, and a JACK_HOOK() function. Both
1363 interfaces create an input and output JACK port. The application makes
1364 these ports the endpoint of the call. The audio coming from the channel
1365 goes out the output port and whatever comes back in on the input port is
1366 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
1367 audiohook on the channel. This lets you run the audio coming from a
1368 channel through JACK, and whatever comes back in is what gets forwarded
1369 on as the channel's audio. This is very useful for building custom
1370 vocoders or doing recording or analysis of the channel's audio in another
1372 * Added a new module, res_config_curl, which permits using a HTTP POST url
1373 to retrieve, create, update, and delete realtime information from a remote
1374 web server. Note that this module requires func_curl.so to be loaded for
1375 backend functionality.
1376 * Added a new module, res_config_ldap, which permits the use of an LDAP
1377 server for realtime data access.
1378 * Added support for writing and running your dialplan in lua using the pbx_lua
1379 module. See configs/extensions.lua.sample for examples of how to do this.
1383 * Ability to use libcap to set high ToS bits when non-root
1384 on Linux. If configure is unable to find libcap then you
1385 can use --with-cap to specify the path.
1386 * Added maxfiles option to options section of asterisk.conf which allows you to specify
1387 what Asterisk should set as the maximum number of open files when it loads.
1388 * Added the jittertargetextra configuration option.
1389 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
1390 configuration files for the IP channel drivers. The new option is "cos".
1391 This information is also documented in doc/qos.tex, or the IP Quality of Service
1392 section of asterisk.pdf.
1393 * When originating a call using AMI or pbx_spool that fails the reason for failure
1394 will now be available in the failed extension using the REASON dialplan variable.
1395 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
1396 It allows you to configure a prefix for auto-monitor recordings.
1397 * A new extension pattern matching algorithm, based on a trie, is introduced
1398 here, that could noticeably speed up mid-sized to large dialplans.
1399 It is NOT used by default, as duplicating the behaviour of the old pattern
1400 matcher is still under development. A config file option, in extensions.conf,
1401 in the [general] section, called "extenpatternmatchingnew", is by default
1402 set to false; setting that to true will force the use of the new algorithm.
1403 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
1404 be used to switch the algorithms at run time.
1405 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
1406 specifying which socket to use to connect to the running Asterisk daemon
1408 * Performance enhancements to the sched facility, which is used in
1409 the channel drivers, etc. Added hashtabs and doubly-linked lists
1410 to speed up deletion; start at the beginning or end of list to
1412 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
1413 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
1414 Added regression tests to the tests/ dir, also.
1415 * Added a refcount trace feature to astobj2 for those trying to balance
1416 object creation, deletion; work, play; space and time. See the
1417 notes in astobj2.h. Also, see utils/refcounter as well, as a
1418 quick way to find unbalanced refcounts in what could be a sea
1419 of objects that were balanced.
1420 * Added logging to 'make update' command. See update.log
1421 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
1422 do not come from the remote party.
1423 * Added the 'n' option to the SpeechBackground application to tell it to not
1424 answer the channel if it has not already been answered.
1425 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
1426 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
1428 * iLBC source code no longer included (see UPGRADE.txt for details)
1429 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
1430 deadlock is detected, a backtrace of the stack which led to the lock calls
1431 will be output to the CLI.
1432 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
1433 the "core show locks" CLI command will give lock information output as well
1434 as a backtrace of the stack which led to the lock calls.
1435 * users.conf now sports an optional alternateexts property, which permits
1436 allocation of additional extensions which will reach the specified user.
1437 * A new option for the configure script, --enable-internal-poll, has been added
1438 for use with systems which may have a buggy implementation of the poll system
1439 call. If you notice odd behavior such as the CLI being unresponsive on remote
1440 consoles, you may want to try using this option. This option is enabled by default
1441 on Darwin systems since it is known that the Darwin poll() implementation has
1445 --------------------
1446 * In addition to timing from DAHDI, there is a new timing module called
1447 res_timing_timerfd. In order to use this, you must be running Linux with
1448 a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
1449 script will be able to tell if you have the requirements. From menuselect, select
1450 res_timing_timerfd from the Resource Modules menu.
projects / asterisk/asterisk.git / blob