-
Notifications
You must be signed in to change notification settings - Fork 0
/
CHANGES
1967 lines (1841 loc) · 110 KB
/
CHANGES
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
619
620
621
622
623
624
625
626
627
628
629
630
631
632
633
634
635
636
637
638
639
640
641
642
643
644
645
646
647
648
649
650
651
652
653
654
655
656
657
658
659
660
661
662
663
664
665
666
667
668
669
670
671
672
673
674
675
676
677
678
679
680
681
682
683
684
685
686
687
688
689
690
691
692
693
694
695
696
697
698
699
700
701
702
703
704
705
706
707
708
709
710
711
712
713
714
715
716
717
718
719
720
721
722
723
724
725
726
727
728
729
730
731
732
733
734
735
736
737
738
739
740
741
742
743
744
745
746
747
748
749
750
751
752
753
754
755
756
757
758
759
760
761
762
763
764
765
766
767
768
769
770
771
772
773
774
775
776
777
778
779
780
781
782
783
784
785
786
787
788
789
790
791
792
793
794
795
796
797
798
799
800
801
802
803
804
805
806
807
808
809
810
811
812
813
814
815
816
817
818
819
820
821
822
823
824
825
826
827
828
829
830
831
832
833
834
835
836
837
838
839
840
841
842
843
844
845
846
847
848
849
850
851
852
853
854
855
856
857
858
859
860
861
862
863
864
865
866
867
868
869
870
871
872
873
874
875
876
877
878
879
880
881
882
883
884
885
886
887
888
889
890
891
892
893
894
895
896
897
898
899
900
901
902
903
904
905
906
907
908
909
910
911
912
913
914
915
916
917
918
919
920
921
922
923
924
925
926
927
928
929
930
931
932
933
934
935
936
937
938
939
940
941
942
943
944
945
946
947
948
949
950
951
952
953
954
955
956
957
958
959
960
961
962
963
964
965
966
967
968
969
970
971
972
973
974
975
976
977
978
979
980
981
982
983
984
985
986
987
988
989
990
991
992
993
994
995
996
997
998
999
1000
==============================================================================
===
=== This file documents the new and/or enhanced functionality added in
=== the Asterisk versions listed below. This file does NOT include
=== changes in behavior that would not be backwards compatible with
=== previous versions; for that information see the UPGRADE.txt file
=== and the other UPGRADE files for older releases.
===
==============================================================================
------------------------------------------------------------------------------
--- Functionality changes since Asterisk 10.1.0 ------------------------------
------------------------------------------------------------------------------
Followme changes
-------------
* A new option, 'I' has been added to app_followme.
By setting this option, Asterisk will not update the caller with
connected line changes when they occur. This is similar to app_dial
and app_queue.
* The 'N' option is now ignored if the call is already answered.
RTP changes
-------------
* A new option, 'probation' has been added to rtp.conf
RTP in strictrtp mode can now require more than 1 packet to exit learning
mode with a new source (and by default requires 4). The probation option
allows the user to change the required number of packets in sequence to any
desired value. Use a value of 1 to essentially restore the old behavior.
Also, with strictrtp on, Asterisk will now drop all packets until learning
mode has successfully exited. These changes are based on how pjmedia handles
media sources and source changes.
Text Messaging
--------------
* MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
instead of simply the uri. This is the format that MessageSend() can use
in the from parameter for outgoing SIP messages.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
------------------------------------------------------------------------------
Text Messaging
--------------
* Asterisk now has protocol independent support for processing text messages
outside of a call. Messages are routed through the Asterisk dialplan.
SIP MESSAGE and XMPP are currently supported. There are options in
jabber.conf and sip.conf to allow enabling these features.
-> jabber.conf: see the "sendtodialplan" and "context" options.
-> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
and "outofcall_message_context" options.
The MESSAGE() dialplan function and MessageSend() application have been
added to go along with this functionality. More detailed usage information
can be found on the Asterisk wiki (http://wiki.asterisk.org/).
* If real-time text support (T.140) is negotiated, it will be preferred for
sending text via the SendText application. For example, via SIP, messages
that were once sent via the SIP MESSAGE request would be sent via RTP if
T.140 text is negotiated for a call.
Parking
-------
* parkedmusicclass can now be set for non-default parking lots.
Asterisk Manager Interface
--------------------------
* PeerStatus now includes Address and Port.
* Added Hold events for when the remote party puts the call on and off hold
for chan_dahdi ISDN channels.
* Added new action MeetmeListRooms to list active conferences (shows same
data as "meetme list" at the CLI).
* DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
Description field that is set by 'description' in the channel configuration
file.
* Added Uniqueid header to UserEvent.
* Added new action FilterAdd to control event filters for the current session.
This requires the system permission and uses the same filter syntax as
filters that can be defined in manager.conf
* The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
versions had some instances of the event converted, but others were left
as-is. All Unlink events should now be converted to Bridge events. The AMI
protocol version number was incremented to 1.2 as a result of this change.
Asterisk HTTP Server
--------------------------
* The HTTP Server can bind to IPv6 addresses.
chan_dahdi
--------------------------
* Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
with busydetect. usage example: busypattern=200,200,200,600
CLI Changes
--------------------------
* New 'gtalk show settings' command showing the current settings loaded from
gtalk.conf.
* The 'logger reload' command now supports an optional argument, specifying an
alternate configuration file to use.
* 'dialplan add extension' command will now automatically create a context if
the specified context does not exist with a message indicated it did so.
* 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
Description field which can be populated with 'description' in the channel
configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
CDR
--------------------------
* The filter option in cdr_adaptive_odbc now supports negating the argument,
thus allowing records which do NOT match the specified filter.
CODECS
--------------------------
* Ability to define custom SILK formats in codecs.conf.
* Addition of speex32 audio format with translation.
* CELT codec pass-through support and ability to define
custom CELT formats in codecs.conf.
* Ability to read raw signed linear files with sample rates
ranging from 8khz - 192khz. The new file extensions introduced
are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
* Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
Skinny, H.323, etc) can still only support the following codecs:
Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
Video: h261, h263, h263p, h264, mpeg4
Image: jpeg, png
Text: red, t140
ConfBridge
--------------------------
* New highly optimized and customizable ConfBridge application capable of
mixing audio at sample rates ranging from 8khz-96khz.
* CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
and bridge profiles on a channel.
* CONFBRIDGE_INFO dialplan function capable of retrieving information
about a conference such as locked status and number of parties, admins,
and marked users.
* Addition of video_mode option in confbridge.conf for adding video support
into a bridge profile.
* Addition of the follow_talker video_mode in confbridge.conf. This video
mode dynamically switches the video feed to always display the loudest talker
supplying video in the conference.
Dialplan Variables
------------------
* Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
variables from asterisk.conf.
Dialplan Functions
------------------
* Addition of the JITTERBUFFER dialplan function. This function allows
for jitterbuffering to occur on the read side of a channel. By using
this function conference applications such as ConfBridge and MeetMe can
have the rx streams jitterbuffered before conference mixing occurs.
* Added DB_KEYS, which lists the next set of keys in the Asterisk database
hierarchy.
* Added STRREPLACE function. This function let's the user search a variable
for a given string to replace with another string as many times as the
user specifies or just throughout the whole string.
* Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
* Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
* Added extensions to chan_ooh323 in function CHANNEL()
libpri channel driver (chan_dahdi) DAHDI changes
--------------------------
* Added moh_signaling option to specify what to do when the channel's bridged
peer puts the ISDN channel on hold.
* Added display_send and display_receive options to control how the display ie
is handled. To send display text from the dialplan use the SendText()
application when the option is enabled.
* Added mcid_send option to allow sending a MCID request on a span.
Calendaring
--------------------------
* Added setvar option to calendar.conf to allow setting channel variables on
notification channels.
* Added "calendar show types" CLI command to list registered calendar
connectors.
MixMonitor
--------------------------
* Added two new options, r and t with file name arguments to record
single direction (unmixed) audio recording separate from the bidirectional
(mixed) recording. The mixed file name argument is optional now as long
as at least one recording option is used.
FollowMe
--------------------------
* Added a new option, l, which will disable local call optimization for
channels involved with the FollowMe thread. Use this option to improve
compatability for a FollowMe call with certain dialplan apps, options, and
functions.
CEL
--------------------------
* cel_pgsql now supports the 'extra' column for data added using the
CELGenUserEvent() application.
pbx_lua
--------------------------
* Support for defining hints has been added to pbx_lua. See the 'hints' table
in the sample extensions.lua file for syntax details.
* Applications that perform jumps in the dialplan such as Goto will now
execute properly. When pbx_lua detects that the context, extension, or
priority we are executing on has changed it will immediately return control
to the asterisk PBX engine. Currently the engine cannot detect a Goto to
the priority after the currently executing priority.
* An autoservice is now started by default for pbx_lua channels. It can be
stopped and restarted using the autoservice_stop() and autoservice_start()
functions.
res_fax
--------------------------
* The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
into a FAXStatus event with an 'Operation' header that will be either
'send', 'receive', or 'gateway'.
* T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
feature will handle converting a fax call between an audio T.30 fax terminal
and an IFP T.38 fax terminal.
SIP Changes
-----------
* Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
* Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
* SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
Queue changes
-------------
* Added general option negative_penalty_invalid default off. when set
members are seen as invalid/logged out when there penalty is negative.
for realtime members when set remove from queue will set penalty to -1.
* Added queue option autopausedelay when autopause is enabled it will be
delayed for this number of seconds since last successful call if there
was no prior call the agent will be autopaused immediately.
* Added member option ignorebusy this when set and ringinuse is not
will allow per member control of multiple calls as ringinuse does for
the Queue.
* Added global option check_state_unknown to enforce checking of device state
when the device state is unknown app_queue will see unknown as available.
Applications
------------
* Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
a MeetMe conference
Asterisk Database
-----------------
* The internal Asterisk database has been switched from Berkeley DB 1.86 to
SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
utility in the UTILS section of menuselect. If an existing astdb is found and no
astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
convert an existing astdb to the SQLite3 version automatically at runtime.
Asterisk Modules
----------------
* Modules marked as deprecated are no longer marked as building by default. Enabling
these modules is still available via menuselect.
IAX2 Changes
------------
* authdebug is now disabled by default. To enable this functionaility again
set authdebug = yes in iax.conf.
RTP Changes
-----------
* The rtp.conf setting "strictrtp" is now enabled by default. In previous
releases it was disabled.
PBX Core
--------
* The PBX core previously made a call with a non-existing extension test for
extension s@default and jump there if the extension existed.
This was a bad default behaviour and violated the principle of least surprise.
It has therefore been changed in this release. It may affect some
applications and configurations that rely on this behaviour. Most channel
drivers have avoided this for many releases by testing whether the extension
called exists before starting the PBX and generating a local error.
This behaviour still exists and works as before.
Extension "s" is used when no extension is given in a channel driver,
like immediate answer in DAHDI or calling to a domain with no user part
in a SIP uri.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
------------------------------------------------------------------------------
SIP Changes
-----------
* Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
now defaults to force_rport. It is very important that phones requiring nat=no be
specifically set as such instead of relying on the default setting. If at all
possible, all devices should have nat settings configured in the general section as
opposed to configuring nat per-device.
* Added preferred_codec_only option in sip.conf. This feature limits the joint
codecs sent in response to an INVITE to the single most preferred codec.
* Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
to be used for the outgoing call. It must be one of the codecs configured
for the device.
* Added tlsprivatekey option to sip.conf. This allows a separate .pem file
to be used for holding a private key. If tlsprivatekey is not specified,
tlscertfile is searched for both public and private key.
* Added tlsclientmethod option to sip.conf. This allows the protocol for
outbound client connections to be specified.
* The sendrpid parameter has been expanded to include the options
'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
header to be sent (equivalent to setting sendrpid=yes) and setting
sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
* The 'ignoresdpversion' behavior has been made automatic when the SDP received
is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
since the call will fail if Asterisk does not process the incoming SDP, Asterisk
will accept the SDP even if the SDP version number is not properly incremented,
but will generate a warning in the log indicating that the SIP peer that sent
the SDP should have the 'ignoresdpversion' option set.
* The 'nat' option has now been been changed to have yes, no, force_rport, and
comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
remote side requests it and disables symmetric RTP support. Setting it to
force_rport forces RFC 3581 behavior and disables symmetric RTP support.
Setting it to comedia enables RFC 3581 behavior if the remote side requests it
and enables symmetric RTP support.
* Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
response. This permits the master channel to know how each channel dialled
in a multi-channel setup resolved in an individual way. This carries a
performance penalty and can be disabled in sip.conf using the
'storesipcause' option.
* Added 'externtcpport' and 'externtlsport' options to allow custom port
configuration for the externip and externhost options when tcp or tls is used.
* Added support for message body (stored in content variable) to SIP NOTIFY message
accessible via AMI and CLI.
* Added 'media_address' configuration option which can be used to explicitly specify
the IP address to use in the SDP for media (audio, video, and text) streams.
* Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
received.
* Added 'use_q850_reason' configuration option for generating and parsing
if available Reason: Q.850;cause=<cause code> header. It is implemented
in some gateways for better passing PRI/SS7 cause codes via SIP.
* When dialing SIP peers, a new component may be added to the end of the dialstring
to indicate that a specific remote IP address or host should be used when dialing
the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
* SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
ability to selectively force bridged channels to also be encrypted is also
implemented. Branching in the dialplan can be done based on whether or not
a channel has secure media and/or signaling.
* Added directmediapermit/directmediadeny to limit which peers can send direct media
to each other
* Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
Charge messages to snom phones.
* Added support for G.719 media streams.
* Added support for 16khz signed linear media streams.
* SIP is now able to bind to and communicate with IPv6 addresses. In addition,
RTP has been outfitted with the same abilities.
* Added support for setting the Max-Forwards: header in SIP requests. Setting is
available in device configurations as well as in the dial plan.
* Addition of the 'subscribe_network_change' option for turning on and off
res_stun_monitor module support in chan_sip.
* Addition of the 'auth_options_requests' option for turning on and off
authentication for OPTIONS requests in chan_sip.
IAX2 Changes
-----------
* Added rtsavesysname option into iax.conf to allow the systname to be saved
on realtime updates.
* Added the ability for chan_iax2 to inform the dialplan whether or not
encryption is being used. This interoperates with the SIP SRTP implementation
so that a secure SIP call can be bridged to a secure IAX call when the
dialplan requires bridged channels to be "secure".
* Addition of the 'subscribe_network_change' option for turning on and off
res_stun_monitor module support in chan_iax.
MGCP Changes
------------
* Added ability to preset channel variables on indicated lines with the setvar
configuration option. Also, clearvars=all resets the list of variables back
to none.
* PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
See configs/res_pktccops.conf for more information.
XMPP Google Talk/Jingle changes
-------------------------------
* Added the externip option to gtalk.conf.
* Added the stunaddr option to gtalk.conf which allows for the automatic
retrieval of the external ip from a stun server.
Applications
------------
* Added 'p' option to PickupChan() to allow for picking up channel by the first
match to a partial channel name.
* Added .m3u support for Mp3Player application.
* Added progress option to the app_dial D() option. When progress DTMF is
present, those values are sent immediately upon receiving a PROGRESS message
regardless if the call has been answered or not.
* Added functionality to the app_dial F() option to continue with execution
at the current location when no parameters are provided.
* Added the 'a' option to app_dial to answer the calling channel before any
announcements or macros are executed.
* Modified app_dial to set answertime when the called channel answers even if
the called channel hangs up during playback of an announcement.
* Modified app_dial 'r' option to support an additional parameter to play an
indication tone from indications.conf
* Added c() option to app_chanspy. This option allows custom DTMF to be set
to cycle through the next available channel. By default this is still '*'.
* Added x() option to app_chanspy. This option allows DTMF to be set to
exit the application.
* The Voicemail application has been improved to automatically ignore messages
that only contain silence.
* If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
associated mailbox(es) to be greetings-only.
* The ChanSpy application now has the 'S' option, which makes the application
automatically exit once it hits a point where no more channels are available
to spy on.
* The ChanSpy application also now has the 'E' option, which spies on a single
channel and exits when that channel hangs up.
* The MeetMe application now turns on the DENOISE() function by default, for
each participant. In our tests, this has significantly decreased background
noise (especially noisy data centers).
* Voicemail now permits storage of secrets in a separate file, located in the
spool directory of each individual user. The control for this is located in
the "passwordlocation" option in voicemail.conf. Please see the sample
configuration for more information.
* The ChanIsAvail application now exposes the returned cause code using a separate
variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
* Added 'd' option to app_followme. This option disables the "Please hold"
announcement.
* Added 'y' option to app_record. This option enables a mode where any DTMF digit
received will terminate recording.
* Voicemail now supports per mailbox settings for folders when using IMAP storage.
Previously the folder could only be set per context, but has now been extended
using the imapfolder option.
* Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
* Voicemail now allows the pager date format to be specified separately from the
email date format.
* New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
to allow joining, leaving, and sending text to group chats.
* MeetMe has a new option 'G' to play an announcement before joining a conference.
* Page has a new option 'A(x)' which will playback an announcement simultaneously
to all paged phones (and optionally excluding the caller's one using the new
option 'n') before the call is bridged.
* The 'f' option to Dial has been augmented to take an optional argument. If no
argument is provided, the 'f' option works as it always has. If an argument is
provided, then the connected party information of all outgoing channels created
during the Dial will be set to the argument passed to the 'f' option.
* Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
Gosub on the peer.
* The OSP lookup application adds in/outbound network ID, optional security,
number portability, QoS reporting, destination IP port, custom info and service
type features.
* Added new application VMSayName that will play the recorded name of the voicemail
user if it exists, otherwise will play the mailbox number.
* Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
retrieve state for a particular bridge, where <name> is the conference name
* app_directory now allows exiting at any time using the operator or pound key.
* Voicemail now supports setting a locale per-mailbox.
* Two new applications are provided for declining counting phrases in multiple
languages. See the application notes for SayCountedNoun and SayCountedAdj for
more information.
* Voicemail now runs the externnotify script when pollmailboxes is activated and
notices a change.
* Voicemail now includes rdnis within msgXXXX.txt file.
* Added 'D' command to ExternalIVR full details in doc/externalivr.txt
* ParkedCall and Park can now specify the parking lot to use.
Dialplan Functions
------------------
* SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
over SRV records associated with a specific service. From the CLI, type
'core show function SRVQUERY' and 'core show function SRVRESULT' for more
details on how these may be used.
* PITCH_SHIFT dialplan function added. This function can be used to modify the
pitch of a channel's tx and rx audio streams.
* Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
setting various connected line and redirecting party information.
* CALLERID and CONNECTEDLINE dialplan functions have been extended to
support ISDN subaddressing.
* The CHANNEL() function now supports the "name" and "checkhangup" options.
* For DAHDI channels, the CHANNEL() dialplan function now allows
the dialplan to request changes in the configuration of the active
echo canceller on the channel (if any), for the current call only.
The syntax is:
exten => s,n,Set(CHANNEL(echocan_mode)=off)
The possible values are:
on - normal mode (the echo canceller is actually reinitialized)
off - disabled
fax - FAX/data mode (NLP disabled if possible, otherwise completely
disabled)
voice - voice mode (returns from FAX mode, reverting the changes that
were made when FAX mode was requested)
* Added new dialplan function MASTER_CHANNEL(), which permits retrieving
and setting variables on the channel which created the current channel.
Administrators should take care to avoid naming conflicts, when multiple
channels are dialled at once, especially when used with the Local channel
construct (which all could set variables on the master channel). Usage
of the HASH() dialplan function, with the key set to the name of the slave
channel, is one approach that will avoid conflicts.
* Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
audio in a channel.
* func_odbc now allows multiple row results to be retrieved without using
mode=multirow. If rowlimit is set, then additional rows may be retrieved
from the same query by using the name of the function which retrieved the
first row as an argument to ODBC_FETCH().
* Added JABBER_RECEIVE, which permits receiving XMPP messages from the
dialplan. This function returns the content of the received message.
* Added REPLACE, which searches a given variable name for a set of characters,
then either replaces them with a single character or deletes them.
* Added PASSTHRU, which literally passes the same argument back as its return
value. The intent is to be able to use a literal string argument to
functions that currently require a variable name as an argument.
* HASH-associated variables now can be inherited across channel creation, by
prefixing the name of the hash at assignment with the appropriate number of
underscores, just like variables.
* GROUP_MATCH_COUNT has been improved to allow regex matching on category
* CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
whether or not channels that are bridged to the current channel will be
required to have secure signaling and/or media.
* CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
the current channel has secure signaling and/or media.
* For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
"no_media_path" option.
Returns "0" if there is a B channel associated with the call.
Returns "1" if no B channel is associated with the call. The call is either
on hold or is a call waiting call.
* Added option to dialplan function CDR(), the 'f' option
allows for high resolution times for billsec and duration fields.
* FILE() now supports line-mode and writing.
* Added FIELDNUM(), which returns the 1-based offset of a field in a list.
* FRAME_TRACE(), for tracking internal ast_frames on a channel.
Dialplan Variables
------------------
* Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
* Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
and is set when a dynamic feature is triggered.
* Added PARKINGLOT which can be used with parkeddynamic feature.conf option
to dynamically create a new parking lot matching the value this varible is
set to.
* Added PARKINGDYNAMIC which represents the template parkinglot defined in
features.conf that should be the base for dynamic parkinglots.
* Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
parkinglot should have.
* Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
parkinglot should have.
* Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
should have.
Queue changes
-------------
* Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
timeout has expired.
* Added 'R' option to app_queue. This option stops moh and indicates ringing
to the caller when an Agent's phone is ringing. This can be used to indicate
to the caller that their call is about to be picked up, which is nice when
one has been on hold for an extened period of time.
* A new config option, penaltymemberslimit, has been added to queues.conf.
When set this option will disregard penalty settings when a queue has too
few members.
* A new option, 'I' has been added to both app_queue and app_dial.
By setting this option, Asterisk will not update the caller with
connected line changes or redirecting party changes when they occur.
* A 'relative-periodic-announce' option has been added to queues.conf. When
enabled, this option will cause periodic announce times to be calculated
from the end of announcements rather than from the beginning.
* The autopause option in queues.conf can be passed a new value, "all." The
result is that if a member becomes auto-paused, he will be paused in all
queues for which he is a member, not just the queue that failed to reach
the member.
* Added dialplan function QUEUE_EXISTS to check if a queue exists
* The queue logger now allows events to optionally propagate to a file,
even when realtime logging is turned on. Additionally, realtime logging
supports sending the event arguments to 5 individual fields, although it
will fallback to the previous data definition, if the new table layout is
not found.
mISDN channel driver (chan_misdn) changes
----------------------------------------
* Added display_connected parameter to misdn.conf to put a display string
in the CONNECT message containing the connected name and/or number if
the presentation setting permits it.
* Added display_setup parameter to misdn.conf to put a display string
in the SETUP message containing the caller name and/or number if the
presentation setting permits it.
* Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
indicate the dialplan settings are to be obtained from the asterisk
channel.
* Made misdn.conf parameter callerid accept the "name" <number> format
used by the rest of the system.
* Made use the nationalprefix and internationalprefix misdn.conf
parameters to prefix any received number from the ISDN link if that
number has the corresponding Type-Of-Number. NOTE: This includes
comparing the incoming call's dialed number against the MSN list.
* Added the following new parameters: unknownprefix, netspecificprefix,
subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
received number from the ISDN link if that number has the corresponding
Type-Of-Number.
* Added new dialplan application misdn_command which permits controlling
the CCBS/CCNR functionality.
* Added new dialplan function mISDN_CC which permits retrieval of various
values from an active call completion record.
* For PTP, you should manually send the COLR of the redirected-to party
for an incomming redirected call if the incoming call could experience
further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
set the REDIRECTING(to-pres) to the COLR. A call has been redirected
if the REDIRECTING(from-num) is not empty.
* For outgoing PTP redirected calls, you now need to use the inhibit(i)
option on all of the REDIRECTING statements before dialing the
redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
and the REDIRECTING(from-xxx,i) values. The PTP call will update the
redirecting-to presentation (COLR) when it becomes available.
* Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
information.
thirdparty mISDN enhancements
-----------------------------
mISDN has been modified by Digium, Inc. to greatly expand facility message
support to allow:
* Enhanced COLP support for call diversion and transfer.
* CCBS/CCNR support.
The latest modified mISDN v1.1.x based version is available at:
http://svn.digium.com/svn/thirdparty/mISDN/trunk
http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
Tagged versions of the modified mISDN code are available under:
http://svn.digium.com/svn/thirdparty/mISDN/tags
http://svn.digium.com/svn/thirdparty/mISDNuser/tags
libpri channel driver (chan_dahdi) DAHDI changes
-------------------------------------------
* The channel variable PRIREDIRECTREASON is now just a status variable
and it is also deprecated. Use the REDIRECTING(reason) dialplan function
to read and alter the reason.
* For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
redirected-to party for an incomming redirected call if the incoming call
could experience further redirects. Just set the
REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
to the COLR. A call has been redirected if the REDIRECTING(count) is not
zero.
* For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
use the inhibit(i) option on all of the REDIRECTING statements before
dialing the redirected-to party. You still have to set the
REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
will update the redirecting-to presentation (COLR) when it becomes available.
* Added the ability to ignore calls that are not in a Multiple Subscriber
Number (MSN) list for PTMP CPE interfaces.
* Added dynamic range compression support for dahdi channels. It is
configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
* Added support for ISDN calling and called subaddress with partial support
for connected line subaddress.
* Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
* Added handling of received HOLD/RETRIEVE messages and the optional ability
to transfer a held call on disconnect similar to an analog phone.
* Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
Will reroute/deflect an outgoing call when receive the message.
Can use the DAHDISendCallreroutingFacility to send the message for the
supported switches.
* Added standard location to add options to chan_dahdi dialing:
Dial(DAHDI/g1[/extension[/options]])
Current options:
K(<keypad_digits>)
R Reverse charging indication
* Added Reverse Charging Indication (Collect calls) send/receive option.
Send reverse charging in SETUP message with the chan_dahdi R dialing option.
Dial(DAHDI/g1/extension/R)
Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
(requires latest LibPRI)
* Added ability to send/receive keypad digits in the SETUP message.
Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
(requires latest LibPRI)
* Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
to eliminate tromboned calls. A tromboned call goes out an interface and comes
back into the same interface. Tromboned calls happen because of call routing,
call deflection, call forwarding, and call transfer.
* Added the ability to send and receive ETSI Advice-Of-Charge messages.
* Added the ability to support call waiting calls. (The SETUP has no B channel
assigned.)
* Added Malicious Call ID (MCID) event to the AMI call event class.
* Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
Asterisk Manager Interface
--------------------------
* The Hangup action now accepts a Cause header which may be used to
set the channel's hangup cause.
* sslprivatekey option added to manager.conf and http.conf. Adds the ability
to specify a separate .pem file to hold a private key. By default sslcert
is used to hold both the public and private key.
* Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
for options containing the 'tls' prefix. For example, 'sslenable' is now
'tlsenable'. This has been done in effort to keep ssl and tls options consistent
across all .conf files. All affected sample.conf files have been modified to
reflect this change. Previous options such as 'sslenable' still work,
but options with the 'tls' prefix are preferred.
* Added a MuteAudio AMI action for muting inbound and/or outbound audio
in a channel. (res_mutestream.so)
* The configuration file manager.conf now supports a channelvars option, which
specifies a list of channel variables to include in each channel-oriented
event.
* The redirect command now has new parameters ExtraContext, ExtraExtension,
and ExtraPriority to allow redirecting the second channel to a different
location than the first.
* Added new event "JabberStatus" in the Jabber module to monitor buddies
status.
* Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
in a MixMonitor recording.
* The 'iax2 show peers' output is now similar to the expected output of
'sip show peers'.
* Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
aoc event class.
* Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
AOC-E messages on a channel.
* A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
conform more closely to similar events.
* Added a new eventfilter option per user to allow whitelisting and blacklisting
of events.
* Added optional parkinglot variable for park command.
* Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
if CallerIDNum and CallerIDName headers are also present.
Channel Event Logging
---------------------
* A new interface, CEL, is introduced here. CEL logs single events, much like
the AMI, but it differs from the AMI in that it logs to db backends much
like CDR does; is based on the event subsystem introduced by Russell, and
can share in all its benefits; allows multiple backends to operate like CDR;
is specialized to event data that would be of concern to billing sytems,
like CDR. Backends for logging and accounting calls have been produced,
but a new CDR backend is still in development.
CDR
---
* 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
* Multiple files and formats can now be specified in cdr_custom.conf.
* cdr_syslog has been added which allows CDRs to be written directly to syslog.
See configs/cdr_syslog.conf.sample for more information.
* A 'sequence' field has been added to CDRs which can be combined with
linkedid or uniqueid to uniquely identify a CDR.
* Handling of billsec and duration field has changed. If your table definition
specifies those fields as float,double or similar they will now be logged with
microsecond accuracy instead of a whole integer.
Calendaring for Asterisk
------------------------
* A new set of modules were added supporing calendar integration with Asterisk.
Dialplan functions for reading from and writing to calendars are included,
as well as the ability to execute dialplan logic upon calendar event notifications.
iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
Exchange Server 2007+ with full write and attendee support) are supported (Exchange
2003 support does not support forms-based authentication).
Call Completion Supplementary Services for Asterisk
---------------------------------------------------
* Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
DAHDI/ISDN supports call completion for the following switch types:
EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
See doc/CCSS_architecture.pdf and doc/tex/ccss.tex(asterisk.pdf) for details.
Multicast RTP Support
---------------------
* A new RTP engine and channel driver have been added which supports Multicast RTP.
The channel driver can be used with the Page application to perform multicast RTP
paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
Type can be either basic or linksys.
Destination is the IP address and port for the RTP packets.
Control address is specific to the linksys type and is used for sending the control
packets unique to them.
Security Events Framework
-------------------------
* Asterisk has a new C API for reporting security events. The module res_security_log
sends these events to the "security" logger level. Currently, AMI is the only
Asterisk component that reports security events. However, SIP support will be
coming soon. For more information on the security events framework, see the
"Security Events" chapter of the included documentation - doc/tex/asterisk.pdf.
Fax
---
* A technology independent fax frontend (res_fax) has been added to Asterisk.
* A spandsp based fax backend (res_fax_spandsp) has been added.
* The app_fax module has been deprecated in favor of the res_fax module and
the new res_fax_spandsp backend.
* The SendFAX and ReceiveFAX applications now send their log messages to a
'fax' logger level, instead of to the generic logger levels. To see these
messages, the system's logger.conf file will need to direct the 'fax' logger
level to one or more destinations; the logger.conf.sample file includes an
example of how to do this. Note that if the 'fax' logger level is *not*
directed to at least one destination, log messages generated by these
applications will be lost, and that if the 'fax' logger level is directed to
the console, the 'core set verbose' and 'core set debug' CLI commands will
have no effect on whether the messages appear on the console or not.
Miscellaneous
-------------
* The transmit_silence_during_record option in asterisk.conf.sample has been removed.
Now, in order to enable transmitting silence during record the transmit_silence
option should be used. transmit_silence_during_record remains a valid option, but
defaults to the behavior of the transmit_silence option.
* Addition of the Unit Test Framework API for managing registration and execution
of unit tests with the purpose of verifying the operation of C functions.
* SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
XMPP text messages to the remote JID.
* Modules.conf has a new option - "require" - that marks a module as critical for
the execution of Asterisk.
If one of the required modules fail to load, Asterisk will exit with a return
code set to 2.
* An 'X' option has been added to the asterisk application which enables #exec support.
This allows #exec to be used in asterisk.conf.
* jabber.conf supports a new option auth_policy that toggles auto user registration.
* A new lockconfdir option has been added to asterisk.conf to protect the
configuration directory (/etc/asterisk by default) during reloads.
* The parkeddynamic option has been added to features.conf to enable the creation
of dynamic parkinglots.
* chan_dahdi now supports reporting alarms over AMI either by channel or span via
the reportalarms config option.
* chan_dahdi supports dialing configuring and dialing by device file name.
DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
* A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
False by default. If set, chan_dahdi will ignore failed 'channel' entries.
Handy for the above name-based syntax as it does not depend on
initialization order.
* The Realtime dialplan switch now caches entries for 1 second. This provides a
significant increase in performance (about 3X) for installations using this switchtype.
* Distributed devicestate now supports the use of the XMPP protocol, in addition to
AIS. For more information, please see doc/distributed_devstate-XMPP.txt
* The addition of G.719 pass-through support.
* Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
during device configuration.
* The UNISTIM channel driver (chan_unistim) has been updated to support devices that
have less than 3 lines on the LCD.
* Realtime now supports database failover. See the sample extconfig.conf for details.
* The addition of improved translation path building for wideband codecs. Sample
rate changes during translation are now avoided unless absolutely necessary.
* The addition of the res_stun_monitor module for monitoring and reacting to network
changes while behind a NAT.
CLI Changes
-----------
* The 'core set debug' and 'core set verbose' commands, in previous versions, could
optionally accept a filename, to apply the setting only to the code generated from
that source file when Asterisk was built. However, there are some modules in Asterisk
that are composed of multiple source files, so this did not result in the behavior
that users expected. In this version, 'core set debug' and 'core set verbose'
can optionally accept *module* names instead (with or without the .so extension),
which applies the setting to the entire module specified, regardless of which source
files it was built from.
* New 'manager show settings' command showing the current settings loaded from
manager.conf.
* Added 'all' keyword to the CLI command "channel request hangup" so that you can send
the channel hangup request to all channels.
* Added a "core reload" CLI command that executes a global reload of Asterisk.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
------------------------------------------------------------------------------
SIP Changes
-----------
* Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
Snom phones use this for call pickup of extensions that the phone is
subscribed to.
* Added support for setting the domain in the URI for caller of an
outbound call by using the SIPFROMDOMAIN channel variable.
* Added a new configuration option "remotesecret" for authentication to
remote services. For backwards compatibility, "secret" still has the
same function as before, but now you can configure both a remote secret and a
local secret for mutual authentication.
* If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
the sound will be played to the target of an attended transfer
* Added two new configuration options, "qualifygap" and "qualifypeers", which allow
finer control over how many peers Asterisk will qualify and the gap between them
when all peers need to be qualified at the same time.
* Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
(either globally or for a specific peer), chan_sip will treat any SDP data
it receives as new data and update the media stream accordingly. By
default, Asterisk will only modify the media stream if the SDP session
version received is different from the current SDP session version. This
option is required to interoperate with devices that have non-standard SDP
session version implementations (observed with Microsoft OCS). This option
is disabled by default.
* The parsing of register => lines in sip.conf has been modified to allow a port
to be present in the "user" portion. Please see the sip.conf.sample file for more
information
* Added support for subscribing to MWI on a remote server and making the status available
as a mailbox. Please see the sip.conf.sample file for more information.
* Added a function to remove SIP headers added in the dialplan before the
first INVITE is generated - SIPRemoveHeader()
* Channel variables set with setvar= in a device configuration is now
set both for inbound and outbound calls.
* Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
IAX2 changes
------------
* Added immediate option to iax.conf
* Added forceencryption option to iax.conf
* Added Encryption and Trunk status to manager command "iaxpeers"
Skinny Changes
--------------
* The configuration file now holds separate sections for devices and lines.
Please have a look at configs/skinny.conf.sample and change your skinny.conf
accordingly.
DAHDI Changes
-------------
* chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
support for LibOpenR2. http://www.libopenr2.org/
* The UK option waitfordialtone has been added for use with BT analog
lines.
* Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
is used in conjunction with the 'faxdetect' configuration option. When
'faxbuffers' is used and fax tones are detected, the channel will dynamically
switch to the configured faxbuffers policy. For example, to use 6 buffers
and a 'full' buffer policy for a fax transmission, add:
faxbuffers=>6,full
The faxbuffers configuration will be in affect until the call is torn down.
* Added service message support for 4ESS/5ESS switches.
Dialplan Functions
------------------
* For DAHDI channels, the CHANNEL() dialplan function now
supports changing the channel's buffer policy (for the current
call only), using this syntax:
exten => s,n,Set(CHANNEL(buffers)=6,full)
This would change the channel to the 'full' buffer policy and
6 (six) buffers. Possible options for this setting are the same
as those in chan_dahdi.conf.
* Added a new dialplan function, CURLOPT, which permits setting various
options that may be useful with the CURL dialplan function, such as
cookies, proxies, connection timeouts, passwords, etc.
* Permit the syntax and synopsis fields of the corresponding dialplan
functions to be individually set from func_odbc.conf.
* Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
* func_odbc now may specify an insert query to execute, when the write query
affects 0 rows (usually indicating that no such row exists).
* Added a new dialplan function, LISTFILTER, which permits removing elements
from a set list, by name. Uses the same general syntax as the existing CUT
and FIELDQTY dialplan functions, which also manage lists.
* Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
obtaining realtime data from the dialplan.
* Added LOCAL_PEEK, which allows access to variables in any stack frame within
a subroutine when using the GoSub() and Return() applications.
* Added AUDIOHOOK_INHERIT. For information on its use, please see the output
of "core show function AUDIOHOOK_INHERIT" from the CLI
* Added AES_ENCRYPT. For information on its use, please see the output
of "core show function AES_ENCRYPT" from the CLI
* Added AES_DECRYPT. For information on its use, please see the output
of "core show function AES_DECRYPT" from the CLI
* func_odbc now supports database transactions across multiple queries.
Applications
------------
* Scheduled meetme conferences may now have their end times extended by
using MeetMeAdmin.
* app_authenticate now gives the ability to select a prompt other than
the default.
* app_directory now pays attention to the searchcontexts setting in
voicemail.conf and will look through all contexts, if no context is
specified in the initial argument.
* A new application, Originate, has been introduced, that allows asynchronous
call origination from the dialplan.
* Voicemail now permits setting the emailsubject and emailbody per mailbox,
in addition to the setting in the "general" context.
* Added ConfBridge dialplan application which does conference bridges without
DAHDI. For information on its use, please see the output of
"core show application ConfBridge" from the CLI.
Miscellaneous
-------------
* The Asterisk CLI has a new command, "channel redirect", which is similar in
operation to the AMI Redirect action.
* extensions.conf now allows you to use keyword "same" to define an extension
without actually specifying an extension. It uses exactly the same pattern
as previously used on the last "exten" line. For example:
exten => 123,1,NoOp(something)
same => n,SomethingElse()
* musiconhold.conf classes of type 'files' can now use relative directory paths,
which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
* All deprecated CLI commands are removed from the sourcecode. They are now handled
by the new clialiases module. See cli_aliases.conf.sample file.
* Times within timespecs are now accurate down to the minute. This is a change
from historical Asterisk, which only provided timespecs rounded to the nearest
even (read: evenly divisible by 2) minute mark.
* The realtime switch now supports an option flag, 'p', which disables searches for
pattern matches.
* In addition to a time range and date range, timespecs now accept a 5th optional
argument, timezone. This allows you to perform time checks on alternate
timezones, especially if those daylight savings time ranges vary from your
machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
includes.