2 ; DAHDI Telephony Configuration file
4 ; You need to restart Asterisk to re-configure the DAHDI channel
5 ; CLI> module reload chan_dahdi.so
6 ; will reload the configuration file, but not all configuration options
7 ; are re-configured during a reload (signalling, as well as PRI and
8 ; SS7-related settings cannot be changed on a reload).
10 ; This file documents many configuration variables. Normally unless you know
11 ; what a variable means or that it should be changed, there's no reason to
12 ; un-comment those lines.
14 ; Examples below that are commented out (those lines that begin with a ';' but
15 ; no space afterwards) typically show a value that is not the default value,
16 ; but would make sense under certain circumstances. The default values are
17 ; usually sane. Thus you should typically not touch them unless you know what
18 ; they mean or you know you should change them.
22 ; Trunk groups are used for NFAS connections.
24 ; Group: Defines a trunk group.
25 ; trunkgroup => <trunkgroup>,<dchannel>[,<backup1>...]
27 ; trunkgroup is the numerical trunk group to create
28 ; dchannel is the DAHDI channel which will have the
29 ; d-channel for the trunk.
30 ; backup1 is an optional list of backup d-channels.
32 ;trunkgroup => 1,24,48
35 ; Spanmap: Associates a span with a trunk group
36 ; spanmap => <dahdispan>,<trunkgroup>[,<logicalspan>]
38 ; dahdispan is the DAHDI span number to associate
39 ; trunkgroup is the trunkgroup (specified above) for the mapping
40 ; logicalspan is the logical span number within the trunk group to use.
41 ; if unspecified, no logical span number is used.
54 ; Context for calls. Defaults to 'default'
58 ; Switchtype: Only used for PRI.
60 ; national: National ISDN 2 (default)
61 ; dms100: Nortel DMS100
64 ; euroisdn: EuroISDN (common in Europe)
65 ; ni1: Old National ISDN 1
70 ; MSNs for ISDN spans. Asterisk will listen for the listed numbers on
71 ; incoming calls and ignore any calls not listed.
72 ; Here you can give a comma separated list of numbers or dialplan extension
73 ; patterns. An empty list disables MSN matching to allow any incoming call.
74 ; Only set on PTMP CPE side of ISDN span if needed.
75 ; The default is an empty list.
78 ; Some switches (AT&T especially) require network specific facility IE.
79 ; Supported values are currently 'none', 'sdn', 'megacom', 'tollfreemegacom', 'accunet'
81 ; nsf cannot be changed on a reload.
85 ;service_message_support=yes
86 ; Enable service message support for channel. Must be set after switchtype.
88 ; Dialing options for ISDN (i.e., Dial(DAHDI/g1/exten/options)):
89 ; R Reverse Charge Indication
90 ; Indicate to the called party that the call will be reverse charged.
91 ; K(n) Keypad digits n
92 ; Send out the specified digits as keypad digits.
94 ; PRI Dialplan: The ISDN-level Type Of Number (TON) or numbering plan, used for
95 ; the dialed number. Leaving this as 'unknown' (the default) works for most
96 ; cases. In some very unusual circumstances, you may need to set this to
97 ; 'dynamic' or 'redundant'.
100 ; private: Private ISDN
102 ; national: National ISDN
103 ; international: International ISDN
104 ; dynamic: Dynamically selects the appropriate dialplan using the
106 ; redundant: Same as dynamic, except that the underlying number is not
107 ; changed (not common)
109 ; pridialplan cannot be changed on reload.
112 ; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's
113 ; numbering plan). In North America, the typical use is sending the 10 digit
114 ; callerID number and setting the prilocaldialplan to 'national' (the default).
115 ; Only VERY rarely will you need to change this.
118 ; private: Private ISDN
120 ; national: National ISDN
121 ; international: International ISDN
122 ; from_channel: Use the CALLERID(ton) value from the channel.
123 ; dynamic: Dynamically selects the appropriate dialplan using the
125 ; redundant: Same as dynamic, except that the underlying number is not
126 ; changed (not common)
128 ; prilocaldialplan cannot be changed on reload.
129 ;prilocaldialplan=national
131 ; PRI Connected Line Dialplan: Sets the connected party number's numbering plan.
134 ; private: Private ISDN
136 ; national: National ISDN
137 ; international: International ISDN
138 ; from_channel: Use the CONNECTEDLINE(ton) value from the channel.
139 ; dynamic: Dynamically selects the appropriate dialplan using the
141 ; redundant: Same as dynamic, except that the underlying number is not
142 ; changed (not common)
144 ; pricpndialplan cannot be changed on reload.
145 ;pricpndialplan=from_channel
147 ; pridialplan may be also set at dialtime, by prefixing the dialed number with
148 ; one of the following letters:
152 ; L - Local (Net Specific)
155 ; R - Reserved (should probably never be used but is included for completeness)
157 ; Additionally, you may also set the following NPI bits (also by prefixing the
158 ; dialed string with one of the following letters):
160 ; e - E.163/E.164 (ISDN/telephony)
165 ; r - Reserved (should probably never be used but is included for completeness)
167 ; You may also set the prilocaldialplan in the same way, but by prefixing the
168 ; Caller*ID Number rather than the dialed number.
170 ; Please note that telcos which require this kind of additional manipulation
171 ; of the TON/NPI are *rare*. Most telco PRIs will work fine simply by
172 ; setting pridialplan to unknown or dynamic.
175 ; PRI caller ID prefixes based on the given TON/NPI (dialplan)
176 ; This is especially needed for EuroISDN E1-PRIs
178 ; None of the prefix settings can be changed on reload.
180 ; sample 1 for Germany
181 ;internationalprefix = 00
184 ;privateprefix = 07115678
187 ; sample 2 for Germany
188 ;internationalprefix = +
189 ;nationalprefix = +49
190 ;localprefix = +49711
191 ;privateprefix = +497115678
194 ; PRI resetinterval: sets the time in seconds between restart of unused
195 ; B channels; defaults to 'never'.
197 ;resetinterval = 3600
199 ; Overlap dialing mode (sending overlap digits)
200 ; Cannot be changed on a reload.
202 ; incoming: incoming direction only
203 ; outgoing: outgoing direction only
204 ; no: neither direction
205 ; yes or both: both directions
209 ; Allow inband audio (progress) when a call is DISCONNECTed by the far end of a PRI
211 ;inbanddisconnect=yes
213 ; Allow a held call to be transferred to the active call on disconnect.
214 ; This is useful on BRI PTMP NT lines where an ISDN phone can simulate the
215 ; transfer feature of an analog phone.
217 ;hold_disconnect_transfer=yes
219 ; PRI Out of band indications.
220 ; Enable this to report Busy and Congestion on a PRI using out-of-band
221 ; notification. Inband indication, as used by Asterisk doesn't seem to work
224 ; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT
225 ; inband: Signal Busy/Congestion using in-band tones (default)
227 ; priindication cannot be changed on a reload.
229 ;priindication = outofband
231 ; If you need to override the existing channels selection routine and force all
232 ; PRI channels to be marked as exclusively selected, set this to yes.
234 ; priexclusive cannot be changed on a reload.
239 ; If you need to use the logical channel mapping with your Q.SIG PRI instead
240 ; of the physical mapping you must use the qsigchannelmapping option.
242 ; logical: Use the logical channel mapping
243 ; physical: Use physical channel mapping (default)
245 ;qsigchannelmapping=logical
247 ; If you wish to ignore remote hold indications (and use MOH that is supplied over
248 ; the B channel) enable this option.
250 ;discardremoteholdretrieval=yes
253 ; All of the ISDN timers and counters that are used are configurable. Specify
254 ; the timer name, and its value (in ms for timers).
255 ; K: Layer 2 max number of outstanding unacknowledged I frames (default 7)
256 ; N200: Layer 2 max number of retransmissions of a frame (default 3)
257 ; T200: Layer 2 max time before retransmission of a frame (default 1000 ms)
258 ; T203: Layer 2 max time without frames being exchanged (default 10000 ms)
259 ; T305: Wait for DISCONNECT acknowledge (default 30000 ms)
260 ; T308: Wait for RELEASE acknowledge (default 4000 ms)
261 ; T309: Maintain active calls on Layer 2 disconnection (default 6000 ms)
262 ; EuroISDN: 6000 to 12000 ms, according to (N200 + 1) x T200 + 2s
263 ; May vary in other ISDN standards (Q.931 1993 : 90000 ms)
264 ; T313: Wait for CONNECT acknowledge, CPE side only (default 3000 ms)
266 ; T-RESPONSE: Maximum time to wait for a typical APDU response. (default 4000 ms)
267 ; This is an implementation timer when the standard does not specify one.
268 ; T-ACTIVATE: Request supervision timeout. (default 10000 ms)
269 ; T-RETENTION: Maximum time to wait for user A to activate call-completion. (default 30000 ms)
270 ; Used by ETSI PTP, ETSI PTMP, and Q.SIG as the cc_offer_timer.
271 ; T-CCBS1: T-STATUS timer equivalent for CC user A status. (default 4000 ms)
272 ; T-CCBS2: Maximum time the CCBS service will be active (default 45 min in ms)
273 ; T-CCBS3: Maximum time to wait for user A to respond to user B availability. (default 20000 ms)
274 ; T-CCBS5: Network B CCBS supervision timeout. (default 60 min in ms)
275 ; T-CCBS6: Network A CCBS supervision timeout. (default 60 min in ms)
276 ; T-CCNR2: Maximum time the CCNR service will be active (default 180 min in ms)
277 ; T-CCNR5: Network B CCNR supervision timeout. (default 195 min in ms)
278 ; T-CCNR6: Network A CCNR supervision timeout. (default 195 min in ms)
279 ; CC-T1: Q.SIG CC request supervision timeout. (default 30000 ms)
280 ; CCBS-T2: Q.SIG CCBS supervision timeout. (default 60 min in ms)
281 ; CCNR-T2: Q.SIG CCNR supervision timeout. (default 195 min in ms)
282 ; CC-T3: Q.SIG CC Maximum time to wait for user A to respond to user B availability. (default 30000 ms)
284 ;pritimer => t200,1000
285 ;pritimer => t313,4000
287 ; CC PTMP recall mode:
288 ; specific - Only the CC original party A can participate in the CC callback
289 ; global - Other compatible endpoints on the PTMP line can be party A in the CC callback
291 ; cc_ptmp_recall_mode cannot be changed on a reload.
293 ;cc_ptmp_recall_mode = specific
295 ; CC Q.SIG Party A (requester) retain signaling link option
296 ; retain Require that the signaling link be retained.
297 ; release Request that the signaling link be released.
298 ; do_not_care The responder is free to choose if the signaling link will be retained.
300 ;cc_qsig_signaling_link_req = retain
302 ; CC Q.SIG Party B (responder) retain signaling link option
303 ; retain Prefer that the signaling link be retained.
304 ; release Prefer that the signaling link be released.
306 ;cc_qsig_signaling_link_rsp = retain
308 ; See ccss.conf.sample for more options. The timers described by ccss.conf.sample
309 ; are not used by ISDN for the native protocol since they are defined by the
310 ; standards and set by pritimer above.
312 ; To enable transmission of facility-based ISDN supplementary services (such
313 ; as caller name from CPE over facility), enable this option.
314 ; Cannot be changed on a reload.
316 ;facilityenable = yes
319 ; This option enables Advice of Charge pass-through between the ISDN PRI and
320 ; Asterisk. This option can be set to any combination of 's', 'd', and 'e' which
321 ; represent the different variants of Advice of Charge, AOC-S, AOC-D, and AOC-E.
322 ; Advice of Charge pass-through is currently only supported for ETSI. Since most
323 ; AOC messages are sent on facility messages, the 'facilityenable' option must
324 ; also be enabled to fully support AOC pass-through.
328 ; When this option is enabled, a hangup initiated by the ISDN PRI side of the
329 ; asterisk channel will result in the channel delaying its hangup in an
330 ; attempt to receive the final AOC-E message from its bridge. The delay
331 ; period is configured as one half the T305 timer length. If the channel
332 ; is not bridged the hangup will occur immediatly without delay.
334 ;aoce_delayhangup=yes
336 ; pritimer cannot be changed on a reload.
338 ; Signalling method. The default is "auto". Valid values:
339 ; auto: Use the current value from DAHDI.
343 ; featd: Feature Group D (The fake, Adtran style, DTMF)
344 ; featdmf: Feature Group D (The real thing, MF (domestic, US))
345 ; featdmf_ta: Feature Group D (The real thing, MF (domestic, US)) through
346 ; a Tandem Access point
347 ; featb: Feature Group B (MF (domestic, US))
348 ; fgccama: Feature Group C-CAMA (DP DNIS, MF ANI)
349 ; fgccamamf: Feature Group C-CAMA MF (MF DNIS, MF ANI)
350 ; fxs_ls: FXS (Loop Start)
351 ; fxs_gs: FXS (Ground Start)
352 ; fxs_ks: FXS (Kewl Start)
353 ; fxo_ls: FXO (Loop Start)
354 ; fxo_gs: FXO (Ground Start)
355 ; fxo_ks: FXO (Kewl Start)
356 ; pri_cpe: PRI signalling, CPE side
357 ; pri_net: PRI signalling, Network side
358 ; bri_cpe: BRI PTP signalling, CPE side
359 ; bri_net: BRI PTP signalling, Network side
360 ; bri_cpe_ptmp: BRI PTMP signalling, CPE side
361 ; bri_net_ptmp: BRI PTMP signalling, Network side
362 ; sf: SF (Inband Tone) Signalling
364 ; sf_featd: SF Feature Group D (The fake, Adtran style, DTMF)
365 ; sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US))
366 ; sf_featb: SF Feature Group B (MF (domestic, US))
367 ; e911: E911 (MF) style signalling
368 ; ss7: Signalling System 7
369 ; mfcr2: MFC/R2 Signalling. To specify the country variant see 'mfcr2_variant'
371 ; The following are used for Radio interfaces:
372 ; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the
374 ; fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the
376 ; fxo_rx: Receive audio/COR on an FXO loopstart interface (FXS at the
378 ; fxo_tx: Transmit audio/PTT on an FXO groundstart interface (FXS at
380 ; em_rx: Receive audio/COR on an E&M interface (1-way)
381 ; em_tx: Transmit audio/PTT on an E&M interface (1-way)
382 ; em_txrx: Receive audio/COR AND Transmit audio/PTT on an E&M interface
384 ; em_rxtx: Same as em_txrx (for our dyslexic friends)
385 ; sf_rx: Receive audio/COR on an SF interface (1-way)
386 ; sf_tx: Transmit audio/PTT on an SF interface (1-way)
387 ; sf_txrx: Receive audio/COR AND Transmit audio/PTT on an SF interface
389 ; sf_rxtx: Same as sf_txrx (for our dyslexic friends)
390 ; ss7: Signalling System 7
392 ; signalling of a channel can not be changed on a reload.
396 ; If you have an outbound signalling format that is different from format
397 ; specified above (but compatible), you can specify outbound signalling format,
398 ; (see below). The 'signalling' format specified will be the inbound signalling
399 ; format. If you only specify 'signalling', then it will be the format for
400 ; both inbound and outbound.
402 ; outsignalling can only be one of:
403 ; em, em_e1, em_w, sf, sf_w, sf_featd, sf_featdmf, sf_featb, featd,
404 ; featdmf, featdmf_ta, e911, fgccama, fgccamamf
406 ; outsignalling cannot be changed on a reload.
412 ; For Feature Group D Tandem access, to set the default CIC and OZZ use these
413 ; parameters (Will not be updated on reload):
418 ; A variety of timing parameters can be specified as well
419 ; The default values for those are "-1", which is to use the
420 ; compile-time defaults of the DAHDI kernel modules. The timing
421 ; parameters, (with the standard default from DAHDI):
423 ; prewink: Pre-wink time (default 50ms)
424 ; preflash: Pre-flash time (default 50ms)
425 ; wink: Wink time (default 150ms)
426 ; flash: Flash time (default 750ms)
427 ; start: Start time (default 1500ms)
428 ; rxwink: Receiver wink time (default 300ms)
429 ; rxflash: Receiver flashtime (default 1250ms)
430 ; debounce: Debounce timing (default 600ms)
432 ; None of them will update on a reload.
434 ; How long generated tones (DTMF and MF) will be played on the channel
437 ; This is a global, rather than a per-channel setting. It will not be
438 ; updated on a reload.
442 ; Whether or not to do distinctive ring detection on FXO lines:
444 ;usedistinctiveringdetection=yes
446 ; enable dring detection after caller ID for those countries like Australia
447 ; where the ring cadence is changed *after* the caller ID spill:
449 ;distinctiveringaftercid=yes
451 ; Whether or not to use caller ID:
455 ; Type of caller ID signalling in use
456 ; bell = bell202 as used in US (default)
457 ; v23 = v23 as used in the UK
458 ; v23_jp = v23 as used in Japan
459 ; dtmf = DTMF as used in Denmark, Sweden and Netherlands
460 ; smdi = Use SMDI for caller ID. Requires SMDI to be enabled (usesmdi).
464 ; What signals the start of caller ID
465 ; ring = a ring signals the start (default)
466 ; polarity = polarity reversal signals the start
467 ; polarity_IN = polarity reversal signals the start, for India,
468 ; for dtmf dialtone detection; using DTMF.
469 ; (see doc/India-CID.txt)
470 ; dtmf = causes monitor loop to look for dtmf energy on the
471 ; incoming channel to initate cid acquisition
475 ; When cidstart=dtmf, the energy level on the line used to trigger dtmf cid
476 ; acquisition. This number is compared to the average over a packet of audio
477 ; of the absolute values of 16 bit signed linear samples. The default is set
478 ; to 256. The choice of 256 is arbitrary. The value you should select should
479 ; be high enough to prevent false detections while low enough to insure that
480 ; no dtmf spills are missed.
484 ; Whether or not to hide outgoing caller ID (Override with *67 or *82)
485 ; (If your dialplan doesn't catch it)
489 ; Enable if you need to hide just the name and not the number for legacy PBX use.
490 ; Only applies to PRI channels.
491 ;hidecalleridname=yes
493 ; On UK analog lines, the caller hanging up determines the end of calls. So
494 ; Asterisk hanging up the line may or may not end a call (DAHDI could just as
495 ; easily be re-attaching to a prior incoming call that was not yet hung up).
496 ; This option changes the hangup to wait for a dialtone on the line, before
497 ; marking the line as once again available for use with outgoing calls.
500 ; The following option enables receiving MWI on FXO lines. The default
502 ; The mwimonitor can take the following values
503 ; no - No mwimonitoring occurs. (default)
504 ; yes - The same as specifying fsk
505 ; fsk - the FXO line is monitored for MWI FSK spills
506 ; fsk,rpas - the FXO line is monitored for MWI FSK spills preceded
507 ; by a ring pulse alert signal.
508 ; neon - The fxo line is monitored for the presence of NEON pulses
510 ; When detected, an internal Asterisk MWI event is generated so that any other
511 ; part of Asterisk that cares about MWI state changes is notified, just as if
512 ; the state change came from app_voicemail.
513 ; For FSK MWI Spills, the energy level that must be seen before starting the
514 ; MWI detection process can be set with 'mwilevel'.
519 ; This option is used in conjunction with mwimonitor. This will get executed
520 ; when incoming MWI state changes. The script is passed 2 arguments. The
521 ; first is the corresponding mailbox, and the second is 1 or 0, indicating if
522 ; there are messages waiting or not.
524 ;mwimonitornotify=/usr/local/bin/dahdinotify.sh
526 ; The following keyword 'mwisendtype' enables various VMWI methods on FXS lines (if supported).
527 ; The default is to send FSK only.
528 ; The following options are available;
529 ; 'rpas' Ring Pulse Alert Signal, alerts intelligent phones that a FSK message is about to be sent.
530 ; 'lrev' Line reversed to indicate messages waiting.
531 ; 'hvdc' 90Vdc OnHook DC voltage to indicate messages waiting.
532 ; 'hvac' or 'neon' 90Vac OnHook AC voltage to light Neon bulb.
533 ; 'nofsk' Disables FSK MWI spills from being sent out.
534 ; It is feasible that multiple options can be enabled.
535 ;mwisendtype=rpas,lrev
537 ; Whether or not to enable call waiting on internal extensions
538 ; With this set to 'yes', busy extensions will hear the call-waiting
539 ; tone, and can use hook-flash to switch between callers. The Dial()
540 ; app will not return the "BUSY" result for extensions.
544 ; Configure the number of outstanding call waiting calls for internal ISDN
545 ; endpoints before bouncing the calls as busy. This option is equivalent to
546 ; the callwaiting option for analog ports.
547 ; A call waiting call is a SETUP message with no B channel selected.
548 ; The default is zero to disable call waiting for ISDN endpoints.
549 ;max_call_waiting_calls=0
551 ; Allow incoming ISDN call waiting calls.
552 ; A call waiting call is a SETUP message with no B channel selected.
553 ;allow_call_waiting_calls=no
555 ; Configure the ISDN span to indicate MWI for the list of mailboxes.
556 ; You can give a comma separated list of up to 8 mailboxes per span.
557 ; An empty list disables MWI.
558 ; The default is an empty list.
559 ;mwi_mailboxes=mailbox_number[@context]{,mailbox_number[@context]}
561 ; Whether or not restrict outgoing caller ID (will be sent as ANI only, not
562 ; available for the user)
563 ; Mostly use with FXS ports
564 ; Does nothing. Use hidecallerid instead.
568 ; Whether or not to use the caller ID presentation from the Asterisk channel
569 ; for outgoing calls.
570 ; See dialplan function CALLERID(pres) for more information.
571 ; Only applies to PRI and SS7 channels.
575 ; Some countries (UK) have ring tones with different ring tones (ring-ring),
576 ; which means the caller ID needs to be set later on, and not just after
577 ; the first ring, as per the default (1).
579 ;sendcalleridafter = 2
582 ; Support caller ID on Call Waiting
584 callwaitingcallerid=yes
586 ; Support three-way calling
590 ; For FXS ports (either direct analog or over T1/E1):
591 ; Support flash-hook call transfer (requires three way calling)
592 ; Also enables call parking (overrides the 'canpark' parameter)
594 ; For digital ports using ISDN PRI protocols:
595 ; Support switch-side transfer (called 2BCT, RLT or other names)
596 ; This setting must be enabled on both ports involved, and the
597 ; 'facilityenable' setting must also be enabled to allow sending
598 ; the transfer to the ISDN switch, since it sent in a FACILITY
600 ; NOTE: This should be disabled for NT PTMP mode. Phones cannot
601 ; have tromboned calls pushed down to them.
606 ; ('canpark=no' is overridden by 'transfer=yes')
610 ; Support call forward variable
614 ; Whether or not to support Call Return (*69, if your dialplan doesn't
619 ; Stutter dialtone support: If a mailbox is specified without a voicemail
620 ; context, then when voicemail is received in a mailbox in the default
621 ; voicemail context in voicemail.conf, taking the phone off hook will cause a
622 ; stutter dialtone instead of a normal one.
624 ; If a mailbox is specified *with* a voicemail context, the same will result
625 ; if voicemail received in mailbox in the specified voicemail context.
627 ; for default voicemail context, the example below is fine:
631 ; for any other voicemail context, the following will produce the stutter tone:
633 ;mailbox=1234@context
635 ; Enable echo cancellation
636 ; Use either "yes", "no", or a power of two from 32 to 256 if you wish to
637 ; actually set the number of taps of cancellation.
639 ; Note that when setting the number of taps, the number 256 does not translate
640 ; to 256 ms of echo cancellation. echocancel=256 means 256 / 8 = 32 ms.
642 ; Note that if any of your DAHDI cards have hardware echo cancellers,
643 ; then this setting only turns them on and off; numeric settings will
644 ; be treated as "yes". There are no special settings required for
645 ; hardware echo cancellers; when present and enabled in their kernel
646 ; modules, they take precedence over the software echo canceller compiled
647 ; into DAHDI automatically.
652 ; Some DAHDI echo cancellers (software and hardware) support adjustable
653 ; parameters; these parameters can be supplied as additional options to
654 ; the 'echocancel' setting. Note that Asterisk does not attempt to
655 ; validate the parameters or their values, so if you supply an invalid
656 ; parameter you will not know the specific reason it failed without
657 ; checking the kernel message log for the error(s) put there by DAHDI.
659 ;echocancel=128,param1=32,param2=0,param3=14
661 ; Generally, it is not necessary (and in fact undesirable) to echo cancel when
662 ; the circuit path is entirely TDM. You may, however, change this behavior
663 ; by enabling the echo canceller during pure TDM bridging below.
665 echocancelwhenbridged=yes
667 ; In some cases, the echo canceller doesn't train quickly enough and there
668 ; is echo at the beginning of the call. Enabling echo training will cause
669 ; DAHDI to briefly mute the channel, send an impulse, and use the impulse
670 ; response to pre-train the echo canceller so it can start out with a much
671 ; closer idea of the actual echo. Value may be "yes", "no", or a number of
672 ; milliseconds to delay before training (default = 400)
674 ; WARNING: In some cases this option can make echo worse! If you are
675 ; trying to debug an echo problem, it is worth checking to see if your echo
676 ; is better with the option set to yes or no. Use whatever setting gives
679 ; Note that these parameters do not apply to hardware echo cancellers.
684 ; If you are having trouble with DTMF detection, you can relax the DTMF
685 ; detection parameters. Relaxing them may make the DTMF detector more likely
686 ; to have "talkoff" where DTMF is detected when it shouldn't be.
690 ; You may also set the default receive and transmit gains (in dB)
692 ; Gain Settings: increasing / decreasing the volume level on a channel.
693 ; The values are in db (decibells). A positive number
694 ; increases the volume level on a channel, and a
695 ; negavive value decreases volume level.
697 ; Dynamic Range Compression: you can also enable dynamic range compression
698 ; on a channel. This will amplify quiet sounds while leaving
699 ; louder sounds untouched. This is useful in situations where
700 ; a linear gain setting would cause clipping. Acceptable values
701 ; are in the range of 0.0 to around 6.0 with higher values
702 ; causing more compression to be done.
704 ; There are several independent gain settings:
705 ; rxgain: gain for the rx (receive - into Asterisk) channel. Default: 0.0
706 ; txgain: gain for the tx (transmit - out of Asterisk Asterisk) channel.
708 ; cid_rxgain: set the gain just for the caller ID sounds Asterisk
709 ; emits. Default: 5.0 .
710 ; rxdrc: dynamic range compression for the rx channel. Default: 0.0
711 ; txdrc: dynamic range compression for the tx channel. Default: 0.0
719 ; Logical groups can be assigned to allow outgoing roll-over. Groups range
720 ; from 0 to 63, and multiple groups can be specified. By default the
721 ; channel is not a member of any group.
723 ; Note that an explicit empty value for 'group' is invalid, and will not
724 ; override a previous non-empty one. The same applies to callgroup and
725 ; pickupgroup as well.
729 ; Ring groups (a.k.a. call groups) and pickup groups. If a phone is ringing
730 ; and it is a member of a group which is one of your pickup groups, then
731 ; you can answer it by picking up and dialing *8#. For simple offices, just
732 ; make these both the same. Groups range from 0 to 63.
737 ; Channel variable to be set for all calls from this channel
739 ;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
740 ; cause the given audio file to
741 ; be played upon completion of
742 ; an attended transfer.
745 ; Specify whether the channel should be answered immediately or if the simple
746 ; switch should provide dialtone, read digits, etc.
747 ; Note: If immediate=yes the dialplan execution will always start at extension
748 ; 's' priority 1 regardless of the dialed number!
752 ; Specify whether flash-hook transfers to 'busy' channels should complete or
753 ; return to the caller performing the transfer (default is yes).
757 ; Calls will have the party id user tag set to this string value.
761 ; With this set, you can automatically append the MSN of a party
762 ; to the cid_tag. An '_' is used to separate the tag from the MSN.
763 ; Applies to ISDN spans.
766 ; Table of what number is appended:
771 ;append_msn_to_cid_tag=no
773 ; caller ID can be set to "asreceived" or a specific number if you want to
774 ; override it. Note that "asreceived" only applies to trunk interfaces.
775 ; fullname sets just the
777 ; fullname: sets just the name part.
778 ; cid_number: sets just the number part:
782 ;callerid = My Name <2564286000>
783 ; Which can also be written as:
784 ;cid_number = 2564286000
787 ;callerid = asreceived
789 ; should we use the caller ID from incoming call on DAHDI transfer?
791 ;useincomingcalleridondahditransfer = yes
793 ; AMA flags affects the recording of Call Detail Records. If specified
794 ; it may be 'default', 'omit', 'billing', or 'documentation'.
798 ; Channels may be associated with an account code to ease
803 ; ADSI (Analog Display Services Interface) can be enabled on a per-channel
804 ; basis if you have (or may have) ADSI compatible CPE equipment
808 ; SMDI (Simplified Message Desk Interface) can be enabled on a per-channel
809 ; basis if you would like that channel to behave like an SMDI message desk.
810 ; The SMDI port specified should have already been defined in smdi.conf. The
811 ; default port is /dev/ttyS0.
816 ; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D
817 ; etc, it can be useful to perform busy detection either in an effort to
818 ; detect hangup or for detecting busies. This enables listening for
819 ; the beep-beep busy pattern.
823 ; If busydetect is enabled, it is also possible to specify how many busy tones
824 ; to wait for before hanging up. The default is 3, but it might be
825 ; safer to set to 6 or even 8. Mind that the higher the number, the more
826 ; time that will be needed to hangup a channel, but lowers the probability
827 ; that you will get random hangups.
831 ; If busydetect is enabled, it is also possible to specify the cadence of your
832 ; busy signal. In many countries, it is 500msec on, 500msec off. Without
833 ; busypattern specified, we'll accept any regular sound-silence pattern that
834 ; repeats <busycount> times as a busy signal. If you specify busypattern,
835 ; then we'll further check the length of the sound (tone) and silence, which
836 ; will further reduce the chance of a false positive.
840 ; NOTE: In make menuselect, you'll find further options to tweak the busy
841 ; detector. If your country has a busy tone with the same length tone and
842 ; silence (as many countries do), consider enabling the
843 ; BUSYDETECT_COMPARE_TONE_AND_SILENCE option.
845 ; To further detect which hangup tone your telco provider is sending, it is
846 ; useful to use the dahdi_monitor utility to record the audio that main/dsp.c
847 ; is receiving after the caller hangs up.
849 ; For FXS (FXO signalled) ports
850 ; switch the line polarity to signal the connected PBX that an outgoing
851 ; call was answered by the remote party.
852 ; For FXO (FXS signalled) ports
853 ; watch for a polarity reversal to mark when a outgoing call is
854 ; answered by the remote party.
856 ;answeronpolarityswitch=yes
858 ; For FXS (FXO signalled) ports
859 ; switch the line polarity to signal the connected PBX that the current
860 ; call was "hung up" by the remote party
861 ; For FXO (FXS signalled) ports
862 ; In some countries, a polarity reversal is used to signal the disconnect of a
863 ; phone line. If the hanguponpolarityswitch option is selected, the call will
864 ; be considered "hung up" on a polarity reversal.
866 ;hanguponpolarityswitch=yes
868 ; polarityonanswerdelay: minimal time period (ms) between the answer
869 ; polarity switch and hangup polarity switch.
872 ; On trunk interfaces (FXS) it can be useful to attempt to follow the progress
873 ; of a call through RINGING, BUSY, and ANSWERING. If turned on, call
874 ; progress attempts to determine answer, busy, and ringing on phone lines.
875 ; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers,
876 ; so don't count on it being very accurate.
878 ; Few zones are supported at the time of this writing, but may be selected
881 ; progzone also affects the pattern used for buzydetect (unless
882 ; busypattern is set explicitly). The possible values are:
884 ; ca (alias for 'us')
886 ; br (Brazil, alias for 'cr')
889 ; This feature can also easily detect false hangups. The symptoms of this is
890 ; being disconnected in the middle of a call for no reason.
895 ; Set the tonezone. Equivalent of the defaultzone settings in
896 ; /etc/dahdi/system.conf. This sets the tone zone by number.
897 ; Note that you'd still need to load tonezones (loadzone in
898 ; /etc/dahdi/system.conf).
899 ; The default is -1: not to set anything.
900 ;tonezone = 0 ; 0 is US
902 ; FXO (FXS signalled) devices must have a timeout to determine if there was a
903 ; hangup before the line was answered. This value can be tweaked to shorten
904 ; how long it takes before DAHDI considers a non-ringing line to have hungup.
906 ; ringtimeout will not update on a reload.
910 ; For FXO (FXS signalled) devices, whether to use pulse dial instead of DTMF
911 ; Pulse digits from phones (FXS devices, FXO signalling) are always
916 ; For fax detection, uncomment one of the following lines. The default is *OFF*
923 ; When 'faxdetect' is used, one could use 'faxbuffers' to configure the DAHDI
924 ; transmit buffer policy. The default is *OFF*. When this configuration
925 ; option is used, the faxbuffer policy will be used for the life of the call
926 ; after a fax tone is detected. The faxbuffer policy is reverted after the
927 ; call is torn down. The sample below will result in 6 buffers and a full
932 ; This option specifies what to do when the channel's bridged peer puts the
933 ; ISDN channel on hold. Settable per logical ISDN span.
934 ; moh: Generate music-on-hold to the remote party.
935 ; notify: Send hold notification signaling to the remote party.
936 ; For ETSI PTP and ETSI PTMP NT links.
937 ; (The notify setting deprecates the mohinterpret=passthrough setting.)
938 ; hold: Use HOLD/RETRIEVE signaling to release the B channel while on hold.
939 ; For ETSI PTMP TE links.
943 ; This option specifies a preference for which music on hold class this channel
944 ; should listen to when put on hold if the music class has not been set on the
945 ; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
946 ; channel putting this one on hold did not suggest a music class.
948 ; This option may be set globally or on a per-channel basis.
950 ;mohinterpret=default
952 ; This option specifies which music on hold class to suggest to the peer channel
953 ; when this channel places the peer on hold. This option may be set globally,
954 ; or on a per-channel basis.
958 ; PRI channels can have an idle extension and a minunused number. So long as
959 ; at least "minunused" channels are idle, chan_dahdi will try to call "idledial"
960 ; on them, and then dump them into the PBX in the "idleext" extension (which
961 ; is of the form exten@context). When channels are needed the "idle" calls
962 ; are disconnected (so long as there are at least "minidle" calls still
963 ; running, of course) to make more channels available. The primary use of
964 ; this is to create a dynamic service, where idle channels are bundled through
965 ; multilink PPP, thus more efficiently utilizing combined voice/data services
966 ; than conventional fixed mappings/muxings.
968 ; Those settings cannot be changed on reload.
971 ;idleext=6999@dialout
976 ; ignore_failed_channels: Continue even if some channels failed to configure.
977 ; False by default, as if even a single channel failed to configure, it might
978 ; mean other channels are misplaced and having them work may not be a good
979 ; idea. If enabled (set to true), chan_dahdi will nevertheless attempt to
980 ; configure other channels rather than giving up. This normally makes sense
981 ; only if you use names (<subdir>!<number>) for DAHDI channels.
982 ;ignore_failed_channels = true
984 ; Configure jitter buffers in DAHDI (each one is 20ms, default is 4)
985 ; This is set globally, rather than per-channel.
989 ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
990 ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
991 ; DAHDI channel. Defaults to "no". An enabled jitterbuffer will
992 ; be used only if the sending side can create and the receiving
993 ; side can not accept jitter. The DAHDI channel can't accept jitter,
994 ; thus an enabled jitterbuffer on the receive DAHDI side will always
995 ; be used if the sending side can create jitter.
997 ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
999 ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
1000 ; resynchronized. Useful to improve the quality of the voice, with
1001 ; big jumps in/broken timestamps, usually sent from exotic devices
1002 ; and programs. Defaults to 1000.
1004 ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a DAHDI
1005 ; channel. Two implementations are currently available - "fixed"
1006 ; (with size always equals to jbmax-size) and "adaptive" (with
1007 ; variable size, actually the new jb of IAX2). Defaults to fixed.
1009 ; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set.
1010 ; The option represents the number of milliseconds by which the new
1011 ; jitter buffer will pad its size. the default is 40, so without
1012 ; modification, the new jitter buffer will set its size to the jitter
1013 ; value plus 40 milliseconds. increasing this value may help if your
1014 ; network normally has low jitter, but occasionally has spikes.
1016 ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
1017 ;-----------------------------------------------------------------------------------
1019 ; You can define your own custom ring cadences here. You can define up to 8
1020 ; pairs. If the silence is negative, it indicates where the caller ID spill is
1021 ; to be placed. Also, if you define any custom cadences, the default cadences
1022 ; will be turned off.
1024 ; This setting is global, rather than per-channel. It will not update on
1027 ; Syntax is: cadence=ring,silence[,ring,silence[...]]
1029 ; These are the default cadences:
1031 ;cadence=125,125,2000,-4000
1032 ;cadence=250,250,500,1000,250,250,500,-4000
1033 ;cadence=125,125,125,125,125,-4000
1034 ;cadence=1000,500,2500,-5000
1036 ; Each channel consists of the channel number or range. It inherits the
1037 ; parameters that were specified above its declaration.
1040 ;callerid="Green Phone"<(256) 428-6121>
1042 ;callerid="Black Phone"<(256) 428-6122>
1044 ;callerid="CallerID Phone" <(630) 372-1564>
1046 ;callerid="Pac Tel Phone" <(256) 428-6124>
1048 ;callerid="Uniden Dead" <(256) 428-6125>
1050 ;callerid="Cortelco 2500" <(256) 428-6126>
1052 ;callerid="Main TA 750" <(256) 428-6127>
1055 ; For example, maybe we have some other channels which start out in a
1056 ; different context and use E & M signalling instead.
1065 ; All those in group 0 I'll use for outgoing calls
1067 ; Strip most significant digit (9) before sending
1070 ;callerid=asreceived
1077 ;callerid="Joe Schmoe" <(256) 428-6131>
1079 ;callerid="Megan May" <(256) 428-6132>
1081 ;callerid="Suzy Queue" <(256) 428-6233>
1083 ;callerid="Larry Moe" <(256) 428-6234>
1086 ; Sample PRI (CPE) config: Specify the switchtype, the signalling as either
1087 ; pri_cpe or pri_net for CPE or Network termination, and generally you will
1088 ; want to create a single "group" for all channels of the PRI.
1090 ; switchtype cannot be changed on a reload.
1092 ; switchtype = national
1093 ; signalling = pri_cpe
1097 ; Alternatively, the number of the channel may be replaced with a relative
1098 ; path to a device file under /dev/dahdi . The final element of that file
1099 ; must be a number, though. The directory separator is '!', as we can't
1100 ; use '/' in a dial string. So if we have
1102 ; /dev/dahdi/span-name/pstn/00/1
1103 ; /dev/dahdi/span-name/pstn/00/2
1104 ; /dev/dahdi/span-name/pstn/00/3
1105 ; /dev/dahdi/span-name/pstn/00/4
1108 ;channel => span-name!pstn!00!1-4
1111 ;channel => span-name!pstn!00!1,2,3,4
1113 ; See also ignore_failed_channels above.
1115 ; Used for distinctive ring support for x100p.
1116 ; You can see the dringX patterns is to set any one of the dringXcontext fields
1117 ; and they will be printed on the console when an inbound call comes in.
1119 ; dringXrange is used to change the acceptable ranges for "tone offsets". Defaults to 10.
1120 ; Note: a range of 0 is NOT what you might expect - it instead forces it to the default.
1121 ; A range of -1 will force it to always match.
1122 ; Anything lower than -1 would presumably cause it to never match.
1125 ;dring1context=internal1
1128 ;dring2context=internal2
1130 ; If no pattern is matched here is where we go.
1134 ; AMI alarm event reporting
1135 ;reportalarms=channels
1136 ;Possible values are:
1137 ;channels - report each channel alarms (current behavior, default for backward compatibility)
1138 ;spans - report an "SpanAlarm" event when the span of any configured channel is alarmed
1139 ;all - report channel and span alarms (aggregated behavior)
1140 ;none - do not report any alarms.
1142 ; ---------------- Options for use with signalling=ss7 -----------------
1143 ; None of them can be changed by a reload.
1145 ; Variant of SS7 signalling:
1146 ; Options are itu and ansi
1149 ; SS7 Called Nature of Address Indicator
1152 ; subscriber: Subscriber
1153 ; national: National
1154 ; international: International
1155 ; dynamic: Dynamically selects the appropriate dialplan
1157 ;ss7_called_nai=dynamic
1159 ; SS7 Calling Nature of Address Indicator
1162 ; subscriber: Subscriber
1163 ; national: National
1164 ; international: International
1165 ; dynamic: Dynamically selects the appropriate dialplan
1167 ;ss7_calling_nai=dynamic
1170 ; sample 1 for Germany
1171 ;ss7_internationalprefix = 00
1172 ;ss7_nationalprefix = 0
1173 ;ss7_subscriberprefix =
1174 ;ss7_unknownprefix =
1177 ; This option is used to disable automatic sending of ACM when the call is started
1178 ; in the dialplan. If you do use this option, you will need to use the Proceeding()
1179 ; application in the dialplan to send ACM.
1182 ; All settings apply to linkset 1
1185 ; Point code of the linkset. For ITU, this is the decimal number
1186 ; format of the point code. For ANSI, this can either be in decimal
1187 ; number format or in the xxx-xxx-xxx format
1190 ; Point code of node adjacent to this signalling link (Possibly the STP between you and
1191 ; your destination). Point code format follows the same rules as above.
1194 ; Default point code that you would like to assign to outgoing messages (in case of
1195 ; routing through STPs, or using A links). Point code format follows the same rules
1199 ; Begin CIC (Circuit indication codes) count with this number
1202 ; What the MTP3 network indicator bits should be set to. Choices are
1203 ; national, national_spare, international, international_spare
1204 ;networkindicator=international
1206 ; First signalling channel
1209 ; Additional signalling channel for this linkset (So you can have a linkset
1210 ; with two signalling links in it). It seems like a silly way to do it, but
1211 ; for linksets with multiple signalling links, you add an additional sigchan
1212 ; line for every additional signalling link on the linkset.
1215 ; Channels to associate with CICs on this linkset
1218 ; For more information on setting up SS7, see the README file in libss7 or
1219 ; the doc/ss7.txt file in the Asterisk source tree.
1220 ; ----------------- SS7 Options ----------------------------------------
1222 ; ---------------- Options for use with signalling=mfcr2 --------------
1224 ; MFC-R2 signaling has lots of variants from country to country and even sometimes
1225 ; minor variants inside the same country. The only mandatory parameters here are:
1226 ; mfcr2_variant, mfcr2_max_ani and mfcr2_max_dnis.
1227 ; IT IS RECOMMENDED that you leave the default values (leaving it commented) for the
1228 ; other parameters unless you have problems or you have been instructed to change some
1229 ; parameter. OpenR2 library uses the mfcr2_variant parameter to try to determine the
1230 ; best defaults for your country, also refer to the OpenR2 package directory
1231 ; doc/asterisk/ where you can find sample configurations for some countries. If you
1232 ; want to contribute your configs for a particular country send them to the e-mail
1233 ; of the primary OpenR2 developer that you can find in the AUTHORS file of the OpenR2 package
1235 ; MFC/R2 variant. This depends on the OpenR2 supported variants
1236 ; A list of values can be found by executing the openr2 command r2test -l
1237 ; some valid values are:
1242 ; itu (per ITU spec)
1245 ; Max amount of ANI to ask for
1248 ; Max amount of DNIS to ask for
1251 ; whether or not to get the ANI before getting DNIS.
1252 ; some telcos require ANI first some others do not care
1253 ; if this go wrong, change this value
1254 ; mfcr2_get_ani_first=no
1256 ; Caller Category to send
1257 ; national_subscriber
1258 ; national_priority_subscriber
1259 ; international_subscriber
1260 ; international_priority_subscriber
1262 ; usually national_subscriber works just fine
1263 ; you can change this setting from the dialplan
1264 ; by setting the variable MFCR2_CATEGORY
1265 ; (remember to set _MFCR2_CATEGORY from originating channels)
1266 ; MFCR2_CATEGORY will also be a variable available in your context
1267 ; on incoming calls set to the value received from the far end
1268 ; mfcr2_category=national_subscriber
1270 ; Call logging is stored at the Asterisk
1271 ; logging directory specified in asterisk.conf
1272 ; plus mfcr2/<whatever you put here>
1273 ; if you specify 'span1' here and asterisk.conf has
1274 ; as logging directory /var/log/asterisk then the full
1275 ; path to your MFC/R2 call logs will be /var/log/asterisk/mfcr2/span1
1276 ; (the directory will be automatically created if not present already)
1277 ; remember to set mfcr2_call_files=yes
1278 ; mfcr2_logdir=span1
1280 ; whether or not to drop call files into mfcr2_logdir
1281 ; mfcr2_call_files=yes|no
1283 ; MFC/R2 valid logging values are: all,error,warning,debug,notice,cas,mf,stack,nothing
1284 ; error,warning,debug and notice are self-descriptive
1285 ; 'cas' is for logging ABCD CAS tx and rx
1286 ; 'mf' is for logging of the Multi Frequency tones
1287 ; 'stack' is for very verbose output of the channel and context call stack, only useful
1288 ; if you are debugging a crash or want to learn how the library works. The stack logging
1289 ; will be only enabled if the openr2 library was compiled with -DOR2_TRACE_STACKS
1290 ; You can mix up values, like: loglevel=error,debug,mf to log just error, debug and
1291 ; multi frequency messages
1292 ; 'all' is a special value to log all the activity
1293 ; 'nothing' is a clean-up value, in case you want to not log any activity for
1294 ; a channel or group of channels
1295 ; BE AWARE that the level of output logged will ALSO depend on
1296 ; the value you have in logger.conf, if you disable output in logger.conf
1297 ; then it does not matter you specify 'all' here, nothing will be logged
1298 ; so logger.conf has the last word on what is going to be logged
1301 ; MFC/R2 value in milliseconds for the MF timeout. Any negative value
1302 ; means 'default', smaller values than 500ms are not recommended
1303 ; and can cause malfunctioning. If you experience protocol error
1304 ; due to MF timeout try incrementing this value in 500ms steps
1305 ; mfcr2_mfback_timeout=-1
1307 ; MFC/R2 value in milliseconds for the metering pulse timeout.
1308 ; Metering pulses are sent by some telcos for some R2 variants
1309 ; during a call presumably for billing purposes to indicate costs,
1310 ; however this pulses use the same signal that is used to indicate
1311 ; call hangup, therefore a timeout is sometimes required to distinguish
1312 ; between a *real* hangup and a billing pulse that should not
1313 ; last more than 500ms, If you experience call drops after some
1314 ; minutes of being stablished try setting a value of some ms here,
1315 ; values greater than 500ms are not recommended.
1316 ; BE AWARE that choosing the proper protocol mfcr2_variant parameter
1317 ; implicitly sets a good recommended value for this timer, use this
1318 ; parameter only when you *really* want to override the default, otherwise
1319 ; just comment out this value or put a -1
1320 ; Any negative value means 'default'.
1321 ; mfcr2_metering_pulse_timeout=-1
1323 ; Brazil uses a special calling party category for collect calls (llamadas por cobrar)
1324 ; instead of using the operator (as in Mexico). The R2 spec in Brazil says a special GB tone
1325 ; should be used to reject collect calls. If you want to ALLOW collect calls specify 'yes',
1326 ; if you want to BLOCK collect calls then say 'no'. Default is to block collect calls.
1327 ; (see also 'mfcr2_double_answer')
1328 ; mfcr2_allow_collect_calls=no
1330 ; This feature is related but independent of mfcr2_allow_collect_calls
1331 ; Some PBX's require a double-answer process to block collect calls, if
1332 ; you ever have problems blocking collect calls using Group B signals (mfcr2_allow_collect_calls=no)
1333 ; then you may want to try with mfcr2_double_answer=yes, this will cause that every answer signal
1334 ; is changed by answer->clear back->answer (sort of a flash)
1335 ; (see also 'mfcr2_allow_collect_calls')
1336 ; mfcr2_double_answer=no
1338 ; This feature allows to skip the use of Group B/II signals and go directly
1339 ; to the accepted state for incoming calls
1340 ; mfcr2_immediate_accept=no
1342 ; You most likely dont need this feature. Default is yes.
1343 ; When this is set to yes, all calls that are offered (incoming calls) which
1344 ; DNIS is valid (exists in extensions.conf) and pass collect call validation
1345 ; will be accepted with a Group B tone (either call with charge or not, depending on mfcr2_charge_calls)
1346 ; with this set to 'no' then the call will NOT be accepted on offered, and the call will start its
1347 ; execution in extensions.conf without being accepted until the channel is answered (either with Answer() or
1348 ; any other application resulting in the channel being answered).
1349 ; This can be set to 'no' if your telco or PBX needs the hangup cause to be set accurately
1350 ; when this option is set to no you must explicitly accept the call with DAHDIAcceptR2Call
1351 ; or implicitly through the Answer() application.
1352 ; mfcr2_accept_on_offer=yes
1354 ; Skip request of calling party category and ANI
1355 ; you need openr2 >= 1.2.0 to use this feature
1356 ; mfcr2_skip_category=no
1358 ; WARNING: advanced users only! I really mean it
1359 ; this parameter is commented by default because
1360 ; YOU DON'T NEED IT UNLESS YOU REALLY GROK MFC/R2
1361 ; READ COMMENTS on doc/r2proto.conf in openr2 package
1363 ; mfcr2_advanced_protocol_file=/path/to/r2proto.conf
1365 ; Brazil use a special signal to force the release of the line (hangup) from the
1366 ; backward perspective. When mfcr2_forced_release=no, the normal clear back signal
1367 ; will be sent on hangup, which is OK for all mfcr2 variants I know of, except for
1368 ; Brazilian variant, where the central will leave the line up for several seconds (30, 60)
1369 ; which sometimes is not what people really want. When mfcr2_forced_release=yes, a different
1370 ; signal will be sent to hangup the call indicating that the line should be released immediately
1371 ; mfcr2_forced_release=no
1373 ; Whether or not report to the other end 'accept call with charge'
1374 ; This setting has no effect with most telecos, usually is safe
1375 ; leave the default (yes), but once in a while when interconnecting with
1376 ; old PBXs this may be useful.
1377 ; Concretely this affects the Group B signal used to accept calls
1378 ; The application DAHDIAcceptR2Call can also be used to decide this
1379 ; in the dial plan in a per-call basis instead of doing it here for all calls
1380 ; mfcr2_charge_calls=yes
1382 ; ---------------- END of options to be used with signalling=mfcr2
1384 ; Configuration Sections
1385 ; ~~~~~~~~~~~~~~~~~~~~~~
1386 ; You can also configure channels in a separate chan_dahdi.conf section. In
1387 ; this case the keyword 'channel' is not used. Instead the keyword
1388 ; 'dahdichan' is used (as in users.conf) - configuration is only processed
1389 ; in a section where the keyword dahdichan is used. It will only be
1390 ; processed in the end of the section. Thus the following section:
1397 ; Is somewhat equivalent to the following snippet in the section
1404 ; When starting a new section almost all of the configuration values are
1405 ; copied from their values at the end of the section [channels] in
1406 ; chan_dahdi.conf and [general] in users.conf - one section's configuration
1407 ; does not affect another one's.
1409 ; Instead of letting common configuration values "slide through" you can
1410 ; use configuration templates to easily keep the common part in one
1411 ; place and override where needed.
1418 ;threewaycalling = yes
1421 ;faxdetect = incoming
1425 ;callerid = My Name <501>
1426 ;mailbox = 501@mailboxes
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