-
Notifications
You must be signed in to change notification settings - Fork 932
/
chan_dahdi.conf.sample
1799 lines (1721 loc) · 70.8 KB
/
chan_dahdi.conf.sample
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
;
; DAHDI Telephony Configuration file
;
; You need to restart Asterisk to re-configure the DAHDI channel
; CLI> module reload chan_dahdi.so
; will reload the configuration file, but not all configuration options
; are re-configured during a reload (signalling, as well as PRI and
; SS7-related settings cannot be changed on a reload).
;
; This file documents many configuration variables. Normally unless you know
; what a variable means or that it should be changed, there's no reason to
; un-comment those lines.
;
; Examples below that are commented out (those lines that begin with a ';' but
; no space afterwards) typically show a value that is not the default value,
; but would make sense under certain circumstances. The default values are
; usually sane. Thus you should typically not touch them unless you know what
; they mean or you know you should change them.
[trunkgroups]
;
; Trunk groups are used for NFAS connections.
;
; Group: Defines a trunk group.
; trunkgroup => <trunkgroup>,<dchannel>[,<backup1>...]
;
; trunkgroup is the numerical trunk group to create
; dchannel is the DAHDI channel which will have the
; d-channel for the trunk.
; backup1 is an optional list of backup d-channels.
;
;trunkgroup => 1,24,48
;trunkgroup => 1,24
;
; Spanmap: Associates a span with a trunk group
; spanmap => <dahdispan>,<trunkgroup>[,<logicalspan>]
;
; dahdispan is the DAHDI span number to associate
; trunkgroup is the trunkgroup (specified above) for the mapping
; logicalspan is the logical span number within the trunk group to use.
; if unspecified, no logical span number is used.
;
;spanmap => 1,1,1
;spanmap => 2,1,2
;spanmap => 3,1,3
;spanmap => 4,1,4
[channels]
;
; Default language
;
;language=en
;
; Context for incoming calls. Defaults to 'default'
;
context=public
;
; Switchtype: Only used for PRI.
;
; national: National ISDN 2 (default)
; dms100: Nortel DMS100
; 4ess: AT&T 4ESS
; 5ess: Lucent 5ESS
; euroisdn: EuroISDN (common in Europe)
; ni1: Old National ISDN 1
; qsig: Q.SIG
;
;switchtype=euroisdn
;
; MSNs for ISDN spans. Asterisk will listen for the listed numbers on
; incoming calls and ignore any calls not listed.
; Here you can give a comma separated list of numbers or dialplan extension
; patterns. An empty list disables MSN matching to allow any incoming call.
; Only set on PTMP CPE side of ISDN span if needed.
; The default is an empty list.
;msn=
;
; Some switches (AT&T especially) require network specific facility IE.
; Supported values are currently 'none', 'sdn', 'megacom', 'tollfreemegacom', 'accunet'
;
; nsf cannot be changed on a reload.
;
;nsf=none
;
;service_message_support=yes
; Enable service message support for channel. Must be set after switchtype.
;
; Dialing options for ISDN (i.e., Dial(DAHDI/g1/exten/options)):
; R Reverse Charge Indication
; Indicate to the called party that the call will be reverse charged.
; K(n) Keypad digits n
; Send out the specified digits as keypad digits.
;
; PRI Dialplan: The ISDN-level Type Of Number (TON) or numbering plan, used for
; the dialed number. Leaving this as 'unknown' (the default) works for most
; cases. In some very unusual circumstances, you may need to set this to
; 'dynamic' or 'redundant'.
;
; unknown: Unknown
; private: Private ISDN
; local: Local ISDN
; national: National ISDN
; international: International ISDN
; dynamic: Dynamically selects the appropriate dialplan using the
; prefix settings.
; redundant: Same as dynamic, except that the underlying number is not
; changed (not common)
;
; pridialplan cannot be changed on reload.
;pridialplan=unknown
;
; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's
; numbering plan). In North America, the typical use is sending the 10 digit
; callerID number and setting the prilocaldialplan to 'national' (the default).
; Only VERY rarely will you need to change this.
;
; unknown: Unknown
; private: Private ISDN
; local: Local ISDN
; national: National ISDN
; international: International ISDN
; from_channel: Use the CALLERID(ton) value from the channel.
; dynamic: Dynamically selects the appropriate dialplan using the
; prefix settings.
; redundant: Same as dynamic, except that the underlying number is not
; changed (not common)
;
; prilocaldialplan cannot be changed on reload.
;prilocaldialplan=national
;
; PRI Connected Line Dialplan: Sets the connected party number's numbering plan.
;
; unknown: Unknown
; private: Private ISDN
; local: Local ISDN
; national: National ISDN
; international: International ISDN
; from_channel: Use the CONNECTEDLINE(ton) value from the channel.
; dynamic: Dynamically selects the appropriate dialplan using the
; prefix settings.
; redundant: Same as dynamic, except that the underlying number is not
; changed (not common)
;
; pricpndialplan cannot be changed on reload.
;pricpndialplan=from_channel
;
; pridialplan may be also set at dialtime, by prefixing the dialed number with
; one of the following letters:
; U - Unknown
; I - International
; N - National
; L - Local (Net Specific)
; S - Subscriber
; V - Abbreviated
; R - Reserved (should probably never be used but is included for completeness)
;
; Additionally, you may also set the following NPI bits (also by prefixing the
; dialed string with one of the following letters):
; u - Unknown
; e - E.163/E.164 (ISDN/telephony)
; x - X.121 (Data)
; f - F.69 (Telex)
; n - National
; p - Private
; r - Reserved (should probably never be used but is included for completeness)
;
; You may also set the prilocaldialplan in the same way, but by prefixing the
; Caller*ID Number rather than the dialed number.
; Please note that telcos which require this kind of additional manipulation
; of the TON/NPI are *rare*. Most telco PRIs will work fine simply by
; setting pridialplan to unknown or dynamic.
;
;
; PRI caller ID prefixes based on the given TON/NPI (dialplan)
; This is especially needed for EuroISDN E1-PRIs
;
; None of the prefix settings can be changed on reload.
;
; sample 1 for Germany
;internationalprefix = 00
;nationalprefix = 0
;localprefix = 0711
;privateprefix = 07115678
;unknownprefix =
;
; sample 2 for Germany
;internationalprefix = +
;nationalprefix = +49
;localprefix = +49711
;privateprefix = +497115678
;unknownprefix =
;
; PRI resetinterval: sets the time in seconds between restart of unused
; B channels; defaults to 'never'.
;
;resetinterval = 3600
;
; Enable per ISDN span to force a RESTART on a channel that returns a cause
; code of PRI_CAUSE_REQUESTED_CHAN_UNAVAIL(44). If this option is enabled
; and the reason the peer rejected the call with cause 44 was that the
; channel is stuck in an unavailable state on the peer, then this might
; help release the channel. It is worth noting that the next outgoing call
; Asterisk makes will likely try the same channel again.
;
; NOTE: Sending a RESTART in response to a cause 44 is not required
; (nor prohibited) by the standards and is likely a primitive chan_dahdi
; response to call collisions (glare) and buggy peers. However, there
; are telco switches out there that ignore the RESTART and continue to
; send calls to the channel in the restarting state.
; Default no.
;
;force_restart_unavailable_chans=yes
;
; Assume inband audio may be present when a SETUP ACK message is received.
; Q.931 Section 5.1.3 says that in scenarios with overlap dialing, when a
; dialtone is sent from the network side, progress indicator 8 "Inband info
; now available" MAY be sent to the CPE if no digits were received with
; the SETUP. It is thus implied that the ie is mandatory if digits came
; with the SETUP and dialtone is needed.
; This option should be enabled, when the network sends dialtone and you
; want to hear it, but the network doesn't send the progress indicator when
; needed.
;
; NOTE: For Q.SIG setups this option should be enabled when outgoing overlap
; dialing is also enabled because Q.SIG does not send the progress indicator
; with the SETUP ACK.
; Default no.
;
;inband_on_setup_ack=yes
;
; Assume inband audio may be present when a PROCEEDING message is received.
; Q.931 Section 5.1.2 says the network cannot assume that the CPE side has
; attached to the B channel at this time without explicitly sending the
; progress indicator ie informing the CPE side to attach to the B channel
; for audio. However, some non-compliant ISDN switches send a PROCEEDING
; without the progress indicator ie indicating inband audio is available and
; assume that the CPE device has connected the media path for listening to
; ringback and other messages.
; Default no.
;
;inband_on_proceeding=yes
;
; Overlap dialing mode (sending overlap digits)
; Cannot be changed on a reload.
;
; incoming: incoming direction only
; outgoing: outgoing direction only
; no: neither direction
; yes or both: both directions
;
;overlapdial=yes
; Send/receive ISDN display IE options. The display options are a comma separated
; list of the following options:
;
; block: Do not pass display text data.
; Q.SIG: Default for send/receive.
; ETSI CPE: Default for send.
; name_initial: Use display text in SETUP/CONNECT messages as the party name.
; Default for all other modes.
; name_update: Use display text in other messages (NOTIFY/FACILITY) for COLP name
; update.
; name: Combined name_initial and name_update options.
; text: Pass any unused display text data as an arbitrary display message
; during a call. Sent text goes out in an INFORMATION message.
;
; * Default is an empty string for legacy behavior.
; * The name options are not recommended for Q.SIG since Q.SIG already
; supports names.
; * The send block is the only recommended setting for CPE mode since Q.931 uses
; the display IE only in the network to user direction.
;
; display_send and display_receive cannot be changed on reload.
;
;display_send=
;display_receive=
; Allow sending an ISDN Malicious Caller ID (MCID) request on this span.
; Default disabled
;
;mcid_send=yes
; Send ISDN date/time IE in CONNECT message option. Only valid on NT spans.
;
; no: Do not send date/time IE in CONNECT message.
; date: Send date only.
; date_hh Send date and hour.
; date_hhmm Send date, hour, and minute.
; date_hhmmss Send date, hour, minute, and second.
;
; Default is an empty string which lets libpri pick the default
; date/time IE send policy.
;
;datetime_send=
; Send ISDN conected line information.
;
; block: Do not send any connected line information.
; connect: Send connected line information on initial connect.
; update: Same as connect but also send any updates during a call.
; Updates happen if the call is transferred. (Default)
;
;colp_send=update
; Allow inband audio (progress) when a call is DISCONNECTed by the far end of a PRI
;
;inbanddisconnect=yes
;
; Allow a held call to be transferred to the active call on disconnect.
; This is useful on BRI PTMP NT lines where an ISDN phone can simulate the
; transfer feature of an analog phone.
; The default is no.
;hold_disconnect_transfer=yes
; BRI PTMP layer 1 presence.
; You should normally not need to set this option.
; You may need to set this option if your telco brings layer 1 down when
; the line is idle.
; required: Layer 1 presence required for outgoing calls. (default)
; ignore: Ignore alarms from DAHDI about this span.
; (Layer 1 and 2 will be brought back up for an outgoing call.)
; NOTE: You will not be able to detect physical line problems
; until an outgoing call is attempted and fails.
;
;layer1_presence=ignore
; BRI PTMP layer 2 persistence.
; You should normally not need to set this option.
; You may need to set this option if your telco brings layer 1 down when
; the line is idle.
; <blank>: Use libpri default.
; keep_up: Bring layer 2 back up if peer takes it down.
; leave_down: Leave layer 2 down if peer takes it down. (Libpri default)
; (Layer 2 will be brought back up for an outgoing call.)
;
;layer2_persistence=leave_down
; PRI Out of band indications.
; Enable this to report Busy and Congestion on a PRI using out-of-band
; notification. Inband indication, as used by Asterisk doesn't seem to work
; with all telcos.
;
; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT
; inband: Signal Busy/Congestion using in-band tones (default)
;
; priindication cannot be changed on a reload.
;
;priindication = outofband
;
; If you need to override the existing channels selection routine and force all
; PRI channels to be marked as exclusively selected, set this to yes.
;
; priexclusive cannot be changed on a reload.
;
;priexclusive = yes
;
;
; If you need to use the logical channel mapping with your Q.SIG PRI instead
; of the physical mapping you must use the qsigchannelmapping option.
;
; logical: Use the logical channel mapping
; physical: Use physical channel mapping (default)
;
;qsigchannelmapping=logical
;
; If you wish to ignore remote hold indications (and use MOH that is supplied over
; the B channel) enable this option.
;
;discardremoteholdretrieval=yes
;
; ISDN Timers
; All of the ISDN timers and counters that are used are configurable. Specify
; the timer name, and its value (in ms for timers).
; K: Layer 2 max number of outstanding unacknowledged I frames (default 7)
; N200: Layer 2 max number of retransmissions of a frame (default 3)
; T200: Layer 2 max time before retransmission of a frame (default 1000 ms)
; T203: Layer 2 max time without frames being exchanged (default 10000 ms)
; T305: Wait for DISCONNECT acknowledge (default 30000 ms)
; T308: Wait for RELEASE acknowledge (default 4000 ms)
; T309: Maintain active calls on Layer 2 disconnection (default 6000 ms)
; EuroISDN: 6000 to 12000 ms, according to (N200 + 1) x T200 + 2s
; May vary in other ISDN standards (Q.931 1993 : 90000 ms)
; T313: Wait for CONNECT acknowledge, CPE side only (default 3000 ms)
;
; T-RESPONSE: Maximum time to wait for a typical APDU response. (default 4000 ms)
; This is an implementation timer when the standard does not specify one.
; T-ACTIVATE: Request supervision timeout. (default 10000 ms)
; T-RETENTION: Maximum time to wait for user A to activate call-completion. (default 30000 ms)
; Used by ETSI PTP, ETSI PTMP, and Q.SIG as the cc_offer_timer.
; T-CCBS1: T-STATUS timer equivalent for CC user A status. (default 4000 ms)
; T-CCBS2: Maximum time the CCBS service will be active (default 45 min in ms)
; T-CCBS3: Maximum time to wait for user A to respond to user B availability. (default 20000 ms)
; T-CCBS5: Network B CCBS supervision timeout. (default 60 min in ms)
; T-CCBS6: Network A CCBS supervision timeout. (default 60 min in ms)
; T-CCNR2: Maximum time the CCNR service will be active (default 180 min in ms)
; T-CCNR5: Network B CCNR supervision timeout. (default 195 min in ms)
; T-CCNR6: Network A CCNR supervision timeout. (default 195 min in ms)
; CC-T1: Q.SIG CC request supervision timeout. (default 30000 ms)
; CCBS-T2: Q.SIG CCBS supervision timeout. (default 60 min in ms)
; CCNR-T2: Q.SIG CCNR supervision timeout. (default 195 min in ms)
; CC-T3: Q.SIG CC Maximum time to wait for user A to respond to user B availability. (default 30000 ms)
;
;pritimer => t200,1000
;pritimer => t313,4000
;
; CC PTMP recall mode:
; specific - Only the CC original party A can participate in the CC callback
; global - Other compatible endpoints on the PTMP line can be party A in the CC callback
;
; cc_ptmp_recall_mode cannot be changed on a reload.
;
;cc_ptmp_recall_mode = specific
;
; CC Q.SIG Party A (requester) retain signaling link option
; retain Require that the signaling link be retained.
; release Request that the signaling link be released.
; do_not_care The responder is free to choose if the signaling link will be retained.
;
;cc_qsig_signaling_link_req = retain
;
; CC Q.SIG Party B (responder) retain signaling link option
; retain Prefer that the signaling link be retained.
; release Prefer that the signaling link be released.
;
;cc_qsig_signaling_link_rsp = retain
;
; See ccss.conf.sample for more options. The timers described by ccss.conf.sample
; are not used by ISDN for the native protocol since they are defined by the
; standards and set by pritimer above.
;
; To enable transmission of facility-based ISDN supplementary services (such
; as caller name from CPE over facility), enable this option.
; Cannot be changed on a reload.
;
;facilityenable = yes
;
; This option enables Advice of Charge pass-through between the ISDN PRI and
; Asterisk. This option can be set to any combination of 's', 'd', and 'e' which
; represent the different variants of Advice of Charge, AOC-S, AOC-D, and AOC-E.
; Advice of Charge pass-through is currently only supported for ETSI. Since most
; AOC messages are sent on facility messages, the 'facilityenable' option must
; also be enabled to fully support AOC pass-through.
;
;aoc_enable=s,d,e
;
; When this option is enabled, a hangup initiated by the ISDN PRI side of the
; asterisk channel will result in the channel delaying its hangup in an
; attempt to receive the final AOC-E message from its bridge. The delay
; period is configured as one half the T305 timer length. If the channel
; is not bridged the hangup will occur immediatly without delay.
;
;aoce_delayhangup=yes
; pritimer cannot be changed on a reload.
;
; Signalling method. The default is "auto". Valid values:
; auto: Use the current value from DAHDI.
; em: E & M
; em_e1: E & M E1
; em_w: E & M Wink
; featd: Feature Group D (The fake, Adtran style, DTMF)
; featdmf: Feature Group D (The real thing, MF (domestic, US))
; featdmf_ta: Feature Group D (The real thing, MF (domestic, US)) through
; a Tandem Access point
; featb: Feature Group B (MF (domestic, US))
; fgccama: Feature Group C-CAMA (DP DNIS, MF ANI)
; fgccamamf: Feature Group C-CAMA MF (MF DNIS, MF ANI)
; fxs_ls: FXS (Loop Start)
; fxs_gs: FXS (Ground Start)
; fxs_ks: FXS (Kewl Start)
; fxo_ls: FXO (Loop Start)
; fxo_gs: FXO (Ground Start)
; fxo_ks: FXO (Kewl Start)
; pri_cpe: PRI signalling, CPE side
; pri_net: PRI signalling, Network side
; bri_cpe: BRI PTP signalling, CPE side
; bri_net: BRI PTP signalling, Network side
; bri_cpe_ptmp: BRI PTMP signalling, CPE side
; bri_net_ptmp: BRI PTMP signalling, Network side
; sf: SF (Inband Tone) Signalling
; sf_w: SF Wink
; sf_featd: SF Feature Group D (The fake, Adtran style, DTMF)
; sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US))
; sf_featb: SF Feature Group B (MF (domestic, US))
; e911: E911 (MF) style signalling
; ss7: Signalling System 7
; mfcr2: MFC/R2 Signalling. To specify the country variant see 'mfcr2_variant'
;
; The following are used for Radio interfaces:
; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the
; channel bank)
; fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the
; channel bank)
; fxo_rx: Receive audio/COR on an FXO loopstart interface (FXS at the
; channel bank)
; fxo_tx: Transmit audio/PTT on an FXO groundstart interface (FXS at
; the channel bank)
; em_rx: Receive audio/COR on an E&M interface (1-way)
; em_tx: Transmit audio/PTT on an E&M interface (1-way)
; em_txrx: Receive audio/COR AND Transmit audio/PTT on an E&M interface
; (2-way)
; em_rxtx: Same as em_txrx (for our dyslexic friends)
; sf_rx: Receive audio/COR on an SF interface (1-way)
; sf_tx: Transmit audio/PTT on an SF interface (1-way)
; sf_txrx: Receive audio/COR AND Transmit audio/PTT on an SF interface
; (2-way)
; sf_rxtx: Same as sf_txrx (for our dyslexic friends)
; ss7: Signalling System 7
;
; signalling of a channel can not be changed on a reload.
;
;signalling=fxo_ls
;
; If you have an outbound signalling format that is different from format
; specified above (but compatible), you can specify outbound signalling format,
; (see below). The 'signalling' format specified will be the inbound signalling
; format. If you only specify 'signalling', then it will be the format for
; both inbound and outbound.
;
; outsignalling can only be one of:
; em, em_e1, em_w, sf, sf_w, sf_featd, sf_featdmf, sf_featb, featd,
; featdmf, featdmf_ta, e911, fgccama, fgccamamf
;
; outsignalling cannot be changed on a reload.
;
;signalling=featdmf
;
;outsignalling=featb
;
; For Feature Group D Tandem access, to set the default CIC and OZZ use these
; parameters (Will not be updated on reload):
;
;defaultozz=0000
;defaultcic=303
;
; A variety of timing parameters can be specified as well
; The default values for those are "-1", which is to use the
; compile-time defaults of the DAHDI kernel modules. The timing
; parameters, (with the standard default from DAHDI):
;
; prewink: Pre-wink time (default 50ms)
; preflash: Pre-flash time (default 50ms)
; wink: Wink time (default 150ms)
; flash: Flash time (default 750ms)
; start: Start time (default 1500ms)
; rxwink: Receiver wink time (default 300ms)
; rxflash: Receiver flashtime (default 1250ms)
; debounce: Debounce timing (default 600ms)
;
; None of them will update on a reload.
;
; How long generated tones (DTMF and MF) will be played on the channel
; (in milliseconds).
;
; This is a global, rather than a per-channel setting. It will not be
; updated on a reload.
;
;toneduration=100
;
; Whether or not to do distinctive ring detection on FXO lines:
;
;usedistinctiveringdetection=yes
;
; enable dring detection after caller ID for those countries like Australia
; where the ring cadence is changed *after* the caller ID spill:
;
;distinctiveringaftercid=yes
;
; Whether or not to use caller ID:
;
usecallerid=yes
;
; NOTE: If the CALL_QUALIFIER variable on the channel is set to 1,
; the Stentor Call Qualifier parameter will be sent for Caller ID spills
; using the Multiple Data Message Format (MDMF).
; This is used by capable Caller ID units to activate the
; "LDC" (Long Distance Call) indicator.
; This variable is not automatically set anywhere. You are responsible
; for setting it in the dialplan if you want to activate the indicator,
; and you must have compatible CPE.
;
; Type of caller ID signalling in use
; bell = bell202 as used in US (default)
; v23 = v23 as used in the UK
; v23_jp = v23 as used in Japan
; dtmf = DTMF as used in Denmark, Sweden and Netherlands
; smdi = Use SMDI for caller ID. Requires SMDI to be enabled (usesmdi).
;
;cidsignalling=v23
;
; What signals the start of caller ID
; ring = a ring signals the start (default)
; polarity = polarity reversal signals the start
; polarity_IN = polarity reversal signals the start, for India,
; for dtmf dialtone detection; using DTMF.
; (see https://wiki.asterisk.org/wiki/display/AST/Caller+ID+in+India)
; dtmf = causes monitor loop to look for dtmf energy on the
; incoming channel to initate cid acquisition
;
;cidstart=polarity
;
; When cidstart=dtmf, the energy level on the line used to trigger dtmf cid
; acquisition. This number is compared to the average over a packet of audio
; of the absolute values of 16 bit signed linear samples. The default is set
; to 256. The choice of 256 is arbitrary. The value you should select should
; be high enough to prevent false detections while low enough to insure that
; no dtmf spills are missed.
;
;dtmfcidlevel=256
;
; Whether or not to hide outgoing caller ID (Override with *67 or *82)
; (If your dialplan doesn't catch it)
;
;hidecallerid=yes
;
; Enable if you need to hide just the name and not the number for legacy PBX use.
; Only applies to PRI channels.
;hidecalleridname=yes
;
; On UK analog lines, the caller hanging up determines the end of calls. So
; Asterisk hanging up the line may or may not end a call (DAHDI could just as
; easily be re-attaching to a prior incoming call that was not yet hung up).
; This option changes the hangup to wait for a dialtone on the line, before
; marking the line as once again available for use with outgoing calls.
; Specified in milliseconds, not set by default.
;waitfordialtone=1000
;
; For analog lines, enables Asterisk to use dialtone detection per channel
; if an incoming call was hung up before it was answered. If dialtone is
; detected, the call is hung up.
; no: Disabled. (Default)
; yes: Look for dialtone for 10000 ms after answer.
; <number>: Look for dialtone for the specified number of ms after answer.
; always: Look for dialtone for the entire call. Dialtone may return
; if the far end hangs up first.
;
;dialtone_detect=no
;
; The following option enables receiving MWI on FXO lines. The default
; value is no.
; The mwimonitor can take the following values
; no - No mwimonitoring occurs. (default)
; yes - The same as specifying fsk
; fsk - the FXO line is monitored for MWI FSK spills
; fsk,rpas - the FXO line is monitored for MWI FSK spills preceded
; by a ring pulse alert signal.
; neon - The fxo line is monitored for the presence of NEON pulses
; indicating MWI.
; When detected, an internal Asterisk MWI event is generated so that any other
; part of Asterisk that cares about MWI state changes is notified, just as if
; the state change came from app_voicemail.
; For FSK MWI Spills, the energy level that must be seen before starting the
; MWI detection process can be set with 'mwilevel'.
;
;mwimonitor=no
;mwilevel=512
;
; This option is used in conjunction with mwimonitor. This will get executed
; when incoming MWI state changes. The script is passed 2 arguments. The
; first is the corresponding configured mailbox, and the second is 1 or 0,
; indicating if there are messages waiting or not.
; Note: app_voicemail mailboxes are in the form of mailbox@context.
;
; /usr/local/bin/dahdinotify.sh 501@mailboxes 1
;
;mwimonitornotify=/usr/local/bin/dahdinotify.sh
;
; The following keyword 'mwisendtype' enables various VMWI methods on FXS lines (if supported).
; The default is to send FSK only.
; The following options are available;
; 'rpas' Ring Pulse Alert Signal, alerts intelligent phones that a FSK message is about to be sent.
; 'lrev' Line reversed to indicate messages waiting.
; 'hvdc' 90Vdc OnHook DC voltage to indicate messages waiting.
; 'hvac' or 'neon' 90Vac OnHook AC voltage to light Neon bulb.
; 'nofsk' Disables FSK MWI spills from being sent out.
; It is feasible that multiple options can be enabled.
;mwisendtype=rpas,lrev
;
; Whether or not to enable call waiting on internal extensions
; With this set to 'yes', busy extensions will hear the call-waiting
; tone, and can use hook-flash to switch between callers. The Dial()
; app will not return the "BUSY" result for extensions.
;
callwaiting=yes
;
; Configure the number of outstanding call waiting calls for internal ISDN
; endpoints before bouncing the calls as busy. This option is equivalent to
; the callwaiting option for analog ports.
; A call waiting call is a SETUP message with no B channel selected.
; The default is zero to disable call waiting for ISDN endpoints.
;max_call_waiting_calls=0
;
; Allow incoming ISDN call waiting calls.
; A call waiting call is a SETUP message with no B channel selected.
;allow_call_waiting_calls=no
; Configure the ISDN span to indicate MWI for the list of mailboxes.
; You can give a comma separated list of up to 8 mailboxes per span.
; An empty list disables MWI.
;
; The default is an empty list.
;mwi_mailboxes=vm-mailbox{,vm-mailbox}
; vm-mailbox = Internal voicemail mailbox identifier.
; Note: app_voicemail mailboxes must be in the form of mailbox@context.
;mwi_mailboxes=501@mailboxes,502@mailboxes
; Configure the ISDN mailbox number sent over the span for MWI mailboxes.
; The position of the number in the list corresponds to the position in
; mwi_mailboxes. If either position in mwi_mailboxes or mwi_vm_boxes is
; empty then that position is disabled.
;
; The default is an empty list.
;mwi_vm_boxes=mailbox_number{,mailbox_number}
;mwi_vm_boxes=501,502
; Configure the ISDN span voicemail controlling numbers for MWI mailboxes.
; What number to call for a user to retrieve voicemail messages.
;
; You can give a comma separated list of numbers. The position of the number
; corresponds to the position in mwi_mailboxes. If a position is empty then
; the last number is reused.
;
; For example:
; mwi_vm_numbers=700,,800,,900
; is equivalent to:
; mwi_vm_numbers=700,700,800,800,900,900,900,900
;
; The default is no number.
;mwi_vm_numbers=
; Whether or not restrict outgoing caller ID (will be sent as ANI only, not
; available for the user)
; Mostly use with FXS ports
; Does nothing. Use hidecallerid instead.
;
;restrictcid=no
;
; Whether or not to use the caller ID presentation from the Asterisk channel
; for outgoing calls.
; See dialplan function CALLERID(pres) for more information.
; Only applies to PRI and SS7 channels.
;
usecallingpres=yes
;
; Some countries (UK) have ring tones with different ring tones (ring-ring),
; which means the caller ID needs to be set later on, and not just after
; the first ring, as per the default (1).
;
;sendcalleridafter = 2
;
;
; Support caller ID on Call Waiting
;
callwaitingcallerid=yes
;
; Whether or not to allow users to go on-hook when receiving an incoming call
; without disconnecting it. Users can later resume the call from any phone
; on the same physical phone line (the same DAHDI channel).
; This setting only has an effect on FXS (FXO-signalled) channels where there
; is only a single incoming call to the DAHDI channel, using the Dial application.
; (This is a convenience mechanism to avoid users wishing to resume a conversation
; at a different phone from leaving a phone off the hook, resuming elsewhere,
; and forgetting to restore the original phone on hook afterwards.)
; Default is no.
;
;calledsubscriberheld=yes
;
; Support three-way calling
;
threewaycalling=yes
;
; By default, the three-way dial tone never times out, allowing it to be
; used as a primitive "hold" mechanism. However, if you'd prefer
; to have the dial tone time out to silence, you can use this option
; to time out after the normal first digit timeout to silence.
; Default is 'no'.
;
;threewaysilenthold=no
;
; For FXS ports (either direct analog or over T1/E1):
; Support flash-hook call transfer (requires three way calling)
; Also enables call parking (overrides the 'canpark' parameter)
;
; For digital ports using ISDN PRI protocols:
; Support switch-side transfer (called 2BCT, RLT or other names)
; This setting must be enabled on both ports involved, and the
; 'facilityenable' setting must also be enabled to allow sending
; the transfer to the ISDN switch, since it sent in a FACILITY
; message.
; NOTE: This should be disabled for NT PTMP mode. Phones cannot
; have tromboned calls pushed down to them.
;
transfer=yes
;
; Allow call parking
; ('canpark=no' is overridden by 'transfer=yes')
;
canpark=yes
; Sets the default parking lot for call parking.
; This is setable per channel.
; Parkinglots are configured in features.conf
;
;parkinglot=plaza
;
; Support call forward variable
;
cancallforward=yes
;
; Whether or not to support Call Return (*69, if your dialplan doesn't
; catch this first)
;
callreturn=yes
;
; Stutter dialtone support: If voicemail is received in the mailbox then
; taking the phone off hook will cause a stutter dialtone instead of a
; normal one.
;
; Note: app_voicemail mailboxes must be in the form of mailbox@context.
;
;mailbox=1234@context
;
; Enable echo cancellation
; Use either "yes", "no", or a power of two from 32 to 256 if you wish to
; actually set the number of taps of cancellation.
;
; Note that when setting the number of taps, the number 256 does not translate
; to 256 ms of echo cancellation. echocancel=256 means 256 / 8 = 32 ms.
;
; Note that if any of your DAHDI cards have hardware echo cancellers,
; then this setting only turns them on and off; numeric settings will
; be treated as "yes". There are no special settings required for
; hardware echo cancellers; when present and enabled in their kernel
; modules, they take precedence over the software echo canceller compiled
; into DAHDI automatically.
;
;
echocancel=yes
;
; Some DAHDI echo cancellers (software and hardware) support adjustable
; parameters; these parameters can be supplied as additional options to
; the 'echocancel' setting. Note that Asterisk does not attempt to
; validate the parameters or their values, so if you supply an invalid
; parameter you will not know the specific reason it failed without
; checking the kernel message log for the error(s) put there by DAHDI.
;
;echocancel=128,param1=32,param2=0,param3=14
;
; Generally, it is not necessary (and in fact undesirable) to echo cancel when
; the circuit path is entirely TDM. You may, however, change this behavior
; by enabling the echo canceller during pure TDM bridging below.
;
echocancelwhenbridged=yes
;
; In some cases, the echo canceller doesn't train quickly enough and there
; is echo at the beginning of the call. Enabling echo training will cause
; DAHDI to briefly mute the channel, send an impulse, and use the impulse
; response to pre-train the echo canceller so it can start out with a much
; closer idea of the actual echo. Value may be "yes", "no", or a number of
; milliseconds to delay before training (default = 400)
;
; WARNING: In some cases this option can make echo worse! If you are
; trying to debug an echo problem, it is worth checking to see if your echo
; is better with the option set to yes or no. Use whatever setting gives
; the best results.
;
; Note that these parameters do not apply to hardware echo cancellers.
;
;echotraining=yes
;echotraining=800
;
; If you are having trouble with DTMF detection, you can relax the DTMF
; detection parameters. Relaxing them may make the DTMF detector more likely
; to have "talkoff" where DTMF is detected when it shouldn't be.
;
;relaxdtmf=yes
;
; Hardware gain settings increase/decrease the analog volume level on a channel.
; The values are in db (decibels) and can be adjusted in 0.1 dB increments.
; A positive number increases the volume level on a channel, and a negavive
; value decreases volume level.
;
; Hardware gain settings are only possible on hardware with analog ports
; because the gain is done on the analog side of the analog/digital conversion.
;
; When hardware gains are disabled, Asterisk will NOT touch the gain setting
; already configured in hardware.
;
; hwrxgain: Hardware receive gain for the channel (into Asterisk).
; Default: disabled
; hwtxgain: Hardware transmit gain for the channel (out of Asterisk).
; Default: disabled
;
;hwrxgain=disabled
;hwtxgain=disabled
;hwrxgain=2.0
;hwtxgain=3.0
;
; Software gain settings digitally increase/decrease the volume level on a channel.
; The values are in db (decibels). A positive number increases the volume
; level on a channel, and a negavive value decreases volume level.
;
; Software gains work on the digital side of the analog/digital conversion
; and thus can also work with T1/E1 cards.
;
; rxgain: Software receive gain for the channel (into Asterisk). Default: 0.0
; txgain: Software transmit gain for the channel (out of Asterisk).
; Default: 0.0
;
; cid_rxgain: Add this gain to rxgain when Asterisk expects to receive
; a Caller ID stream.
; Default: 5.0 .
;
;rxgain=2.0
;txgain=3.0
;
; Dynamic Range Compression: You can also enable dynamic range compression
; on a channel. This will digitally amplify quiet sounds while leaving louder
; sounds untouched. This is useful in situations where a linear gain setting
; would cause clipping. Acceptable values are in the range of 0.0 to around
; 6.0 with higher values causing more compression to be done.
;
; rxdrc: dynamic range compression for the rx channel. Default: 0.0
; txdrc: dynamic range compression for the tx channel. Default: 0.0
;
;rxdrc=1.0
;txdrc=4.0
;
; Logical groups can be assigned to allow outgoing roll-over. Groups range
; from 0 to 63, and multiple groups can be specified. By default the
; channel is not a member of any group.
;
; Note that an explicit empty value for 'group' is invalid, and will not
; override a previous non-empty one. The same applies to callgroup and
; pickupgroup as well.
;
group=1
;
; Ring groups (a.k.a. call groups) and pickup groups. If a phone is ringing
; and it is a member of a group which is one of your pickup groups, then
; you can answer it by picking up and dialing *8#. For simple offices, just
; make these both the same. Groups range from 0 to 63.
;
; Call groups and pickup groups may only be specified for FXO signalled channels.
; If you need to pick up an FXS signalled channel directly, you can have it
; dial a Local channel and pick up the ;1 side of the Local channel instead.
;
callgroup=1
pickupgroup=1
;
; Named ring groups (a.k.a. named call groups) and named pickup groups.
; If a phone is ringing and it is a member of a group which is one of your
; named pickup groups, then you can answer it by picking up and dialing *8#.
; For simple offices, just make these both the same.
; The number of named groups is not limited.
;
;namedcallgroup=engineering,sales,netgroup,protgroup
;namedpickupgroup=sales
; Channel variables to be set for all calls from this channel
;setvar=CHANNEL=42
;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
; cause the given audio file to
; be played upon completion of
; an attended transfer to the
; target of the transfer.
;
; On FXS channels (FXO signaled), specifies whether the channel should enter the dialplan
; immediately or if the simple switch should provide dialtone, read digits, etc.
; On FXO channels (FXS signaled), specifies whether the call should enter the dialplan
; immediately or if we should wait for at least one ring. This is required if
; Caller ID or distinctive ringing is enabled. If you do not need either, you can
; skip waiting for the first ring to begin call processing sooner.
;
; Note: If immediate=yes the dialplan execution will always start at extension
; 's' priority 1 regardless of the dialed number!
;
;immediate=yes
;
; On FXS channels (FXO signaled), specifies whether fake audible ringback should
; be provided as soon as the channel goes off hook and immediate=yes.
; If audio should come only from the dialplan, this option should be disabled.
; Default is 'yes'
;
;immediatering=no
;
; Specify whether flash-hook transfers to 'busy' channels should complete or
; return to the caller performing the transfer (default is yes).
;
;transfertobusy=no
; Calls will have the party id user tag set to this string value.
;
;cid_tag=
; With this set, you can automatically append the MSN of a party