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 ; PRI Reverse Charging Indication: Indicate to the called party that the
89 ; call will be reverse charged. To enable, prefix the dialed number with one
90 ; of the following letters:
91 ; C - Reverse Charge Indication Requested
93 ; PRI Dialplan: The ISDN-level Type Of Number (TON) or numbering plan, used for
94 ; the dialed number. For most installations, leaving this as 'unknown' (the
95 ; default) works in the most cases. In some very unusual circumstances, you
96 ; may need to set this to 'dynamic' or 'redundant'. Note that if you set one
97 ; of the others, you will be unable to dial another class of numbers. For
98 ; example, if you set 'national', you will be unable to dial local or
99 ; international numbers.
101 ; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's
102 ; numbering plan). In North America, the typical use is sending the 10 digit
103 ; callerID number and setting the prilocaldialplan to 'national' (the default).
104 ; Only VERY rarely will you need to change this.
106 ; Neither pridialplan nor prilocaldialplan can be changed on reload.
109 ; private: Private ISDN
111 ; national: National ISDN
112 ; international: International ISDN
113 ; dynamic: Dynamically selects the appropriate dialplan
114 ; redundant: Same as dynamic, except that the underlying number is not
115 ; changed (not common)
118 ;prilocaldialplan=national
120 ; pridialplan may be also set at dialtime, by prefixing the dialled number with
121 ; one of the following letters:
125 ; L - Local (Net Specific)
128 ; R - Reserved (should probably never be used but is included for completeness)
130 ; Additionally, you may also set the following NPI bits (also by prefixing the
131 ; dialled string with one of the following letters):
133 ; e - E.163/E.164 (ISDN/telephony)
138 ; r - Reserved (should probably never be used but is included for completeness)
140 ; You may also set the prilocaldialplan in the same way, but by prefixing the
141 ; Caller*ID Number, rather than the dialled number. Please note that telcos
142 ; which require this kind of additional manipulation of the TON/NPI are *rare*.
143 ; Most telco PRIs will work fine simply by setting pridialplan to unknown or
147 ; PRI caller ID prefixes based on the given TON/NPI (dialplan)
148 ; This is especially needed for EuroISDN E1-PRIs
150 ; None of the prefix settings can be changed on reload.
152 ; sample 1 for Germany
153 ;internationalprefix = 00
156 ;privateprefix = 07115678
159 ; sample 2 for Germany
160 ;internationalprefix = +
161 ;nationalprefix = +49
162 ;localprefix = +49711
163 ;privateprefix = +497115678
166 ; PRI resetinterval: sets the time in seconds between restart of unused
167 ; B channels; defaults to 'never'.
169 ;resetinterval = 3600
171 ; Overlap dialing mode (sending overlap digits)
172 ; Cannot be changed on a reload.
174 ; incoming: incoming direction only
175 ; outgoing: outgoing direction only
176 ; no: neither direction
177 ; yes or both: both directions
181 ; Allow inband audio (progress) when a call is DISCONNECTed by the far end of a PRI
183 ;inbanddisconnect=yes
185 ; Allow a held call to be transferred to the active call on disconnect.
186 ; This is useful on BRI PTMP NT lines where an ISDN phone can simulate the
187 ; transfer feature of an analog phone.
189 ;hold_disconnect_transfer=yes
191 ; PRI Out of band indications.
192 ; Enable this to report Busy and Congestion on a PRI using out-of-band
193 ; notification. Inband indication, as used by Asterisk doesn't seem to work
196 ; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT
197 ; inband: Signal Busy/Congestion using in-band tones (default)
199 ; priindication cannot be changed on a reload.
201 ;priindication = outofband
203 ; If you need to override the existing channels selection routine and force all
204 ; PRI channels to be marked as exclusively selected, set this to yes.
206 ; priexclusive cannot be changed on a reload.
211 ; If you need to use the logical channel mapping with your Q.SIG PRI instead
212 ; of the physical mapping you must use the qsigchannelmapping option.
214 ; logical: Use the logical channel mapping
215 ; physical: Use physical channel mapping (default)
217 ;qsigchannelmapping=logical
219 ; If you wish to ignore remote hold indications (and use MOH that is supplied over
220 ; the B channel) enable this option.
222 ;discardremoteholdretrieval=yes
225 ; All of the ISDN timers and counters that are used are configurable. Specify
226 ; the timer name, and its value (in ms for timers).
227 ; K: Layer 2 max number of outstanding unacknowledged I frames (default 7)
228 ; N200: Layer 2 max number of retransmissions of a frame (default 3)
229 ; T200: Layer 2 max time before retransmission of a frame (default 1000 ms)
230 ; T203: Layer 2 max time without frames being exchanged (default 10000 ms)
231 ; T305: Wait for DISCONNECT acknowledge (default 30000 ms)
232 ; T308: Wait for RELEASE acknowledge (default 4000 ms)
233 ; T309: Maintain active calls on Layer 2 disconnection (default -1,
234 ; Asterisk clears calls)
235 ; EuroISDN: 6000 to 12000 ms, according to (N200 + 1) x T200 + 2s
236 ; May vary in other ISDN standards (Q.931 1993 : 90000 ms)
237 ; T313: Wait for CONNECT acknowledge, CPE side only (default 3000 ms)
239 ;pritimer => t200,1000
240 ;pritimer => t313,4000
242 ; To enable transmission of facility-based ISDN supplementary services (such
243 ; as caller name from CPE over facility), enable this option.
244 ; Cannot be changed on a reload.
246 ;facilityenable = yes
248 ; pritimer cannot be changed on a reload.
250 ; Signalling method. The default is "auto". Valid values:
251 ; auto: Use the current value from DAHDI.
255 ; featd: Feature Group D (The fake, Adtran style, DTMF)
256 ; featdmf: Feature Group D (The real thing, MF (domestic, US))
257 ; featdmf_ta: Feature Group D (The real thing, MF (domestic, US)) through
258 ; a Tandem Access point
259 ; featb: Feature Group B (MF (domestic, US))
260 ; fgccama: Feature Group C-CAMA (DP DNIS, MF ANI)
261 ; fgccamamf: Feature Group C-CAMA MF (MF DNIS, MF ANI)
262 ; fxs_ls: FXS (Loop Start)
263 ; fxs_gs: FXS (Ground Start)
264 ; fxs_ks: FXS (Kewl Start)
265 ; fxo_ls: FXO (Loop Start)
266 ; fxo_gs: FXO (Ground Start)
267 ; fxo_ks: FXO (Kewl Start)
268 ; pri_cpe: PRI signalling, CPE side
269 ; pri_net: PRI signalling, Network side
270 ; sf: SF (Inband Tone) Signalling
272 ; sf_featd: SF Feature Group D (The fake, Adtran style, DTMF)
273 ; sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US))
274 ; sf_featb: SF Feature Group B (MF (domestic, US))
275 ; e911: E911 (MF) style signalling
276 ; ss7: Signalling System 7
277 ; mfcr2: MFC/R2 Signalling. To specify the country variant see 'mfcr2_variant'
279 ; The following are used for Radio interfaces:
280 ; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the
282 ; fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the
284 ; fxo_rx: Receive audio/COR on an FXO loopstart interface (FXS at the
286 ; fxo_tx: Transmit audio/PTT on an FXO groundstart interface (FXS at
288 ; em_rx: Receive audio/COR on an E&M interface (1-way)
289 ; em_tx: Transmit audio/PTT on an E&M interface (1-way)
290 ; em_txrx: Receive audio/COR AND Transmit audio/PTT on an E&M interface
292 ; em_rxtx: Same as em_txrx (for our dyslexic friends)
293 ; sf_rx: Receive audio/COR on an SF interface (1-way)
294 ; sf_tx: Transmit audio/PTT on an SF interface (1-way)
295 ; sf_txrx: Receive audio/COR AND Transmit audio/PTT on an SF interface
297 ; sf_rxtx: Same as sf_txrx (for our dyslexic friends)
298 ; ss7: Signalling System 7
300 ; signalling of a channel can not be changed on a reload.
304 ; If you have an outbound signalling format that is different from format
305 ; specified above (but compatible), you can specify outbound signalling format,
306 ; (see below). The 'signalling' format specified will be the inbound signalling
307 ; format. If you only specify 'signalling', then it will be the format for
308 ; both inbound and outbound.
310 ; outsignalling can only be one of:
311 ; em, em_e1, em_w, sf, sf_w, sf_featd, sf_featdmf, sf_featb, featd,
312 ; featdmf, featdmf_ta, e911, fgccama, fgccamamf
314 ; outsignalling cannot be changed on a reload.
320 ; For Feature Group D Tandem access, to set the default CIC and OZZ use these
321 ; parameters (Will not be updated on reload):
326 ; A variety of timing parameters can be specified as well
327 ; The default values for those are "-1", which is to use the
328 ; compile-time defaults of the DAHDI kernel modules. The timing
329 ; parameters, (with the standard default from DAHDI):
331 ; prewink: Pre-wink time (default 50ms)
332 ; preflash: Pre-flash time (default 50ms)
333 ; wink: Wink time (default 150ms)
334 ; flash: Flash time (default 750ms)
335 ; start: Start time (default 1500ms)
336 ; rxwink: Receiver wink time (default 300ms)
337 ; rxflash: Receiver flashtime (default 1250ms)
338 ; debounce: Debounce timing (default 600ms)
340 ; None of them will update on a reload.
342 ; How long generated tones (DTMF and MF) will be played on the channel
345 ; This is a global, rather than a per-channel setting. It will not be
346 ; updated on a reload.
350 ; Whether or not to do distinctive ring detection on FXO lines:
352 ;usedistinctiveringdetection=yes
354 ; enable dring detection after caller ID for those countries like Australia
355 ; where the ring cadence is changed *after* the caller ID spill:
357 ;distinctiveringaftercid=yes
359 ; Whether or not to use caller ID:
363 ; Hide the name part and leave just the number part of the caller ID
364 ; string. Only applies to PRI channels.
365 ;hidecalleridname=yes
367 ; Type of caller ID signalling in use
368 ; bell = bell202 as used in US (default)
369 ; v23 = v23 as used in the UK
370 ; v23_jp = v23 as used in Japan
371 ; dtmf = DTMF as used in Denmark, Sweden and Netherlands
372 ; smdi = Use SMDI for caller ID. Requires SMDI to be enabled (usesmdi).
376 ; What signals the start of caller ID
377 ; ring = a ring signals the start (default)
378 ; polarity = polarity reversal signals the start
379 ; polarity_IN = polarity reversal signals the start, for India,
380 ; for dtmf dialtone detection; using DTMF.
381 ; (see doc/India-CID.txt)
382 ; dtmf = causes monitor loop to look for dtmf energy on the
383 ; incoming channel to initate cid acquisition
387 ; When cidstart=dtmf, the energy level on the line used to trigger dtmf cid
388 ; acquisition. This number is compared to the average over a packet of audio
389 ; of the absolute values of 16 bit signed linear samples. The default is set
390 ; to 256. The choice of 256 is arbitrary. The value you should select should
391 ; be high enough to prevent false detections while low enough to insure that
392 ; no dtmf spills are missed.
396 ; Whether or not to hide outgoing caller ID (Override with *67 or *82)
397 ; (If your dialplan doesn't catch it)
401 ; On UK analog lines, the caller hanging up determines the end of calls. So
402 ; Asterisk hanging up the line may or may not end a call (DAHDI could just as
403 ; easily be re-attaching to a prior incoming call that was not yet hung up).
404 ; This option changes the hangup to wait for a dialtone on the line, before
405 ; marking the line as once again available for use with outgoing calls.
408 ; The following option enables receiving MWI on FXO lines. The default
410 ; The mwimonitor can take the following values
411 ; no - No mwimonitoring occurs. (default)
412 ; yes - The same as specifying fsk
413 ; fsk - the FXO line is monitored for MWI FSK spills
414 ; fsk,rpas - the FXO line is monitored for MWI FSK spills preceded
415 ; by a ring pulse alert signal.
416 ; neon - The fxo line is monitored for the presence of NEON pulses
418 ; When detected, an internal Asterisk MWI event is generated so that any other
419 ; part of Asterisk that cares about MWI state changes is notified, just as if
420 ; the state change came from app_voicemail.
421 ; For FSK MWI Spills, the energy level that must be seen before starting the
422 ; MWI detection process can be set with 'mwilevel'.
427 ; This option is used in conjunction with mwimonitor. This will get executed
428 ; when incoming MWI state changes. The script is passed 2 arguments. The
429 ; first is the corresponding mailbox, and the second is 1 or 0, indicating if
430 ; there are messages waiting or not.
432 ;mwimonitornotify=/usr/local/bin/dahdinotify.sh
434 ; The following keyword 'mwisendtype' enables various VMWI methods on FXS lines (if supported).
435 ; The default is to send FSK only.
436 ; The following options are available;
437 ; 'rpas' Ring Pulse Alert Signal, alerts intelligent phones that a FSK message is about to be sent.
438 ; 'lrev' Line reversed to indicate messages waiting.
439 ; 'hvdc' 90Vdc OnHook DC voltage to indicate messages waiting.
440 ; 'hvac' or 'neon' 90Vac OnHook AC voltage to light Neon bulb.
441 ; 'nofsk' Disables FSK MWI spills from being sent out.
442 ; It is feasible that multiple options can be enabled.
443 ;mwisendtype=rpas,lrev
445 ; Whether or not to enable call waiting on internal extensions
446 ; With this set to 'yes', busy extensions will hear the call-waiting
447 ; tone, and can use hook-flash to switch between callers. The Dial()
448 ; app will not return the "BUSY" result for extensions.
452 ; Whether or not restrict outgoing caller ID (will be sent as ANI only, not
453 ; available for the user)
454 ; Mostly use with FXS ports
455 ; Does nothing. Use hidecallerid instead.
459 ; Whether or not to use the caller ID presentation from the Asterisk channel
460 ; for outgoing calls.
461 ; See dialplan function CALLERID(pres) for more information.
462 ; Only applies to PRI and SS7 channels.
466 ; Some countries (UK) have ring tones with different ring tones (ring-ring),
467 ; which means the caller ID needs to be set later on, and not just after
468 ; the first ring, as per the default (1).
470 ;sendcalleridafter = 2
473 ; Support caller ID on Call Waiting
475 callwaitingcallerid=yes
477 ; Support three-way calling
481 ; For FXS ports (either direct analog or over T1/E1):
482 ; Support flash-hook call transfer (requires three way calling)
483 ; Also enables call parking (overrides the 'canpark' parameter)
485 ; For digital ports using ISDN PRI protocols:
486 ; Support switch-side transfer (called 2BCT, RLT or other names)
487 ; This setting must be enabled on both ports involved, and the
488 ; 'facilityenable' setting must also be enabled to allow sending
489 ; the transfer to the ISDN switch, since it sent in a FACILITY
495 ; ('canpark=no' is overridden by 'transfer=yes')
499 ; Support call forward variable
503 ; Whether or not to support Call Return (*69, if your dialplan doesn't
508 ; Stutter dialtone support: If a mailbox is specified without a voicemail
509 ; context, then when voicemail is received in a mailbox in the default
510 ; voicemail context in voicemail.conf, taking the phone off hook will cause a
511 ; stutter dialtone instead of a normal one.
513 ; If a mailbox is specified *with* a voicemail context, the same will result
514 ; if voicemail received in mailbox in the specified voicemail context.
516 ; for default voicemail context, the example below is fine:
520 ; for any other voicemail context, the following will produce the stutter tone:
522 ;mailbox=1234@context
524 ; Enable echo cancellation
525 ; Use either "yes", "no", or a power of two from 32 to 256 if you wish to
526 ; actually set the number of taps of cancellation.
528 ; Note that when setting the number of taps, the number 256 does not translate
529 ; to 256 ms of echo cancellation. echocancel=256 means 256 / 8 = 32 ms.
531 ; Note that if any of your DAHDI cards have hardware echo cancellers,
532 ; then this setting only turns them on and off; numeric settings will
533 ; be treated as "yes". There are no special settings required for
534 ; hardware echo cancellers; when present and enabled in their kernel
535 ; modules, they take precedence over the software echo canceller compiled
536 ; into DAHDI automatically.
541 ; Some DAHDI echo cancellers (software and hardware) support adjustable
542 ; parameters; these parameters can be supplied as additional options to
543 ; the 'echocancel' setting. Note that Asterisk does not attempt to
544 ; validate the parameters or their values, so if you supply an invalid
545 ; parameter you will not know the specific reason it failed without
546 ; checking the kernel message log for the error(s) put there by DAHDI.
548 ;echocancel=128,param1=32,param2=0,param3=14
550 ; Generally, it is not necessary (and in fact undesirable) to echo cancel when
551 ; the circuit path is entirely TDM. You may, however, change this behavior
552 ; by enabling the echo canceller during pure TDM bridging below.
554 echocancelwhenbridged=yes
556 ; In some cases, the echo canceller doesn't train quickly enough and there
557 ; is echo at the beginning of the call. Enabling echo training will cause
558 ; DAHDI to briefly mute the channel, send an impulse, and use the impulse
559 ; response to pre-train the echo canceller so it can start out with a much
560 ; closer idea of the actual echo. Value may be "yes", "no", or a number of
561 ; milliseconds to delay before training (default = 400)
563 ; WARNING: In some cases this option can make echo worse! If you are
564 ; trying to debug an echo problem, it is worth checking to see if your echo
565 ; is better with the option set to yes or no. Use whatever setting gives
568 ; Note that these parameters do not apply to hardware echo cancellers.
573 ; If you are having trouble with DTMF detection, you can relax the DTMF
574 ; detection parameters. Relaxing them may make the DTMF detector more likely
575 ; to have "talkoff" where DTMF is detected when it shouldn't be.
579 ; You may also set the default receive and transmit gains (in dB)
581 ; Gain Settings: increasing / decreasing the volume level on a channel.
582 ; The values are in db (decibells). A positive number
583 ; increases the volume level on a channel, and a
584 ; negavive value decreases volume level.
586 ; Dynamic Range Compression: you can also enable dynamic range compression
587 ; on a channel. This will amplify quiet sounds while leaving
588 ; louder sounds untouched. This is useful in situations where
589 ; a linear gain setting would cause clipping. Acceptable values
590 ; are in the range of 0.0 to around 6.0 with higher values
591 ; causing more compression to be done.
593 ; There are several independent gain settings:
594 ; rxgain: gain for the rx (receive - into Asterisk) channel. Default: 0.0
595 ; txgain: gain for the tx (transmit - out of Asterisk Asterisk) channel.
597 ; cid_rxgain: set the gain just for the caller ID sounds Asterisk
598 ; emits. Default: 5.0 .
599 ; rxdrc: dynamic range compression for the rx channel. Default: 0.0
600 ; txdrc: dynamic range compression for the tx channel. Default: 0.0
608 ; Logical groups can be assigned to allow outgoing roll-over. Groups range
609 ; from 0 to 63, and multiple groups can be specified. By default the
610 ; channel is not a member of any group.
612 ; Note that an explicit empty value for 'group' is invalid, and will not
613 ; override a previous non-empty one. The same applies to callgroup and
614 ; pickupgroup as well.
618 ; Ring groups (a.k.a. call groups) and pickup groups. If a phone is ringing
619 ; and it is a member of a group which is one of your pickup groups, then
620 ; you can answer it by picking up and dialing *8#. For simple offices, just
621 ; make these both the same. Groups range from 0 to 63.
626 ; Channel variable to be set for all calls from this channel
628 ;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
629 ; cause the given audio file to
630 ; be played upon completion of
631 ; an attended transfer.
634 ; Specify whether the channel should be answered immediately or if the simple
635 ; switch should provide dialtone, read digits, etc.
636 ; Note: If immediate=yes the dialplan execution will always start at extension
637 ; 's' priority 1 regardless of the dialed number!
641 ; Specify whether flash-hook transfers to 'busy' channels should complete or
642 ; return to the caller performing the transfer (default is yes).
646 ; caller ID can be set to "asreceived" or a specific number if you want to
647 ; override it. Note that "asreceived" only applies to trunk interfaces.
648 ; fullname sets just the
650 ; fullname: sets just the name part.
651 ; cid_number: sets just the number part:
655 ;callerid = My Name <2564286000>
656 ; Which can also be written as:
657 ;cid_number = 2564286000
660 ;callerid = asreceived
662 ; should we use the caller ID from incoming call on DAHDI transfer?
664 ;useincomingcalleridondahditransfer = yes
666 ; AMA flags affects the recording of Call Detail Records. If specified
667 ; it may be 'default', 'omit', 'billing', or 'documentation'.
671 ; Channels may be associated with an account code to ease
676 ; ADSI (Analog Display Services Interface) can be enabled on a per-channel
677 ; basis if you have (or may have) ADSI compatible CPE equipment
681 ; SMDI (Simplified Message Desk Interface) can be enabled on a per-channel
682 ; basis if you would like that channel to behave like an SMDI message desk.
683 ; The SMDI port specified should have already been defined in smdi.conf. The
684 ; default port is /dev/ttyS0.
689 ; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D
690 ; etc, it can be useful to perform busy detection either in an effort to
691 ; detect hangup or for detecting busies. This enables listening for
692 ; the beep-beep busy pattern.
696 ; If busydetect is enabled, it is also possible to specify how many busy tones
697 ; to wait for before hanging up. The default is 3, but it might be
698 ; safer to set to 6 or even 8. Mind that the higher the number, the more
699 ; time that will be needed to hangup a channel, but lowers the probability
700 ; that you will get random hangups.
704 ; If busydetect is enabled, it is also possible to specify the cadence of your
705 ; busy signal. In many countries, it is 500msec on, 500msec off. Without
706 ; busypattern specified, we'll accept any regular sound-silence pattern that
707 ; repeats <busycount> times as a busy signal. If you specify busypattern,
708 ; then we'll further check the length of the sound (tone) and silence, which
709 ; will further reduce the chance of a false positive.
713 ; NOTE: In make menuselect, you'll find further options to tweak the busy
714 ; detector. If your country has a busy tone with the same length tone and
715 ; silence (as many countries do), consider enabling the
716 ; BUSYDETECT_COMPARE_TONE_AND_SILENCE option.
718 ; To further detect which hangup tone your telco provider is sending, it is
719 ; useful to use the ztmonitor utility to record the audio that main/dsp.c
720 ; is receiving after the caller hangs up.
722 ; Use a polarity reversal to mark when a outgoing call is answered by the
725 ;answeronpolarityswitch=yes
727 ; In some countries, a polarity reversal is used to signal the disconnect of a
728 ; phone line. If the hanguponpolarityswitch option is selected, the call will
729 ; be considered "hung up" on a polarity reversal.
731 ;hanguponpolarityswitch=yes
733 ; polarityonanswerdelay: minimal time period (ms) between the answer
734 ; polarity switch and hangup polarity switch.
737 ; On trunk interfaces (FXS) it can be useful to attempt to follow the progress
738 ; of a call through RINGING, BUSY, and ANSWERING. If turned on, call
739 ; progress attempts to determine answer, busy, and ringing on phone lines.
740 ; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers,
741 ; so don't count on it being very accurate.
743 ; Few zones are supported at the time of this writing, but may be selected
746 ; progzone also affects the pattern used for buzydetect (unless
747 ; busypattern is set explicitly). The possible values are:
749 ; ca (alias for 'us')
751 ; br (Brazil, alias for 'cr')
754 ; This feature can also easily detect false hangups. The symptoms of this is
755 ; being disconnected in the middle of a call for no reason.
760 ; Set the tonezone. Equivalent of the defaultzone settings in
761 ; /etc/dahdi/system.conf. This sets the tone zone by number.
762 ; Note that you'd still need to load tonezones (loadzone in
763 ; /etc/dahdi/system.conf).
764 ; The default is -1: not to set anything.
765 ;tonezone = 0 ; 0 is US
767 ; FXO (FXS signalled) devices must have a timeout to determine if there was a
768 ; hangup before the line was answered. This value can be tweaked to shorten
769 ; how long it takes before DAHDI considers a non-ringing line to have hungup.
771 ; ringtimeout will not update on a reload.
775 ; For FXO (FXS signalled) devices, whether to use pulse dial instead of DTMF
776 ; Pulse digits from phones (FXS devices, FXO signalling) are always
781 ; For fax detection, uncomment one of the following lines. The default is *OFF*
788 ; When 'faxdetect' is used, one could use 'faxbuffers' to configure the DAHDI
789 ; transmit buffer policy. The default is *OFF*. When this configuration
790 ; option is used, the faxbuffer policy will be used for the life of the call
791 ; after a fax tone is detected. The faxbuffer policy is reverted after the
792 ; call is torn down. The sample below will result in 6 buffers and a full
797 ; This option specifies a preference for which music on hold class this channel
798 ; should listen to when put on hold if the music class has not been set on the
799 ; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
800 ; channel putting this one on hold did not suggest a music class.
802 ; If this option is set to "passthrough", then the hold message will always be
803 ; passed through as signalling instead of generating hold music locally. This
804 ; setting is only valid when used on a channel that uses digital signalling.
806 ; This option may be set globally or on a per-channel basis.
808 ;mohinterpret=default
810 ; This option specifies which music on hold class to suggest to the peer channel
811 ; when this channel places the peer on hold. This option may be set globally,
812 ; or on a per-channel basis.
816 ; PRI channels can have an idle extension and a minunused number. So long as
817 ; at least "minunused" channels are idle, chan_dahdi will try to call "idledial"
818 ; on them, and then dump them into the PBX in the "idleext" extension (which
819 ; is of the form exten@context). When channels are needed the "idle" calls
820 ; are disconnected (so long as there are at least "minidle" calls still
821 ; running, of course) to make more channels available. The primary use of
822 ; this is to create a dynamic service, where idle channels are bundled through
823 ; multilink PPP, thus more efficiently utilizing combined voice/data services
824 ; than conventional fixed mappings/muxings.
826 ; Those settings cannot be changed on reload.
829 ;idleext=6999@dialout
833 ; Configure jitter buffers in DAHDI (each one is 20ms, default is 4)
834 ; This is set globally, rather than per-channel.
838 ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
839 ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
840 ; DAHDI channel. Defaults to "no". An enabled jitterbuffer will
841 ; be used only if the sending side can create and the receiving
842 ; side can not accept jitter. The DAHDI channel can't accept jitter,
843 ; thus an enabled jitterbuffer on the receive DAHDI side will always
844 ; be used if the sending side can create jitter.
846 ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
848 ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
849 ; resynchronized. Useful to improve the quality of the voice, with
850 ; big jumps in/broken timestamps, usually sent from exotic devices
851 ; and programs. Defaults to 1000.
853 ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a DAHDI
854 ; channel. Two implementations are currently available - "fixed"
855 ; (with size always equals to jbmax-size) and "adaptive" (with
856 ; variable size, actually the new jb of IAX2). Defaults to fixed.
858 ; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set.
859 ; The option represents the number of milliseconds by which the new
860 ; jitter buffer will pad its size. the default is 40, so without
861 ; modification, the new jitter buffer will set its size to the jitter
862 ; value plus 40 milliseconds. increasing this value may help if your
863 ; network normally has low jitter, but occasionally has spikes.
865 ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
866 ;-----------------------------------------------------------------------------------
868 ; You can define your own custom ring cadences here. You can define up to 8
869 ; pairs. If the silence is negative, it indicates where the caller ID spill is
870 ; to be placed. Also, if you define any custom cadences, the default cadences
871 ; will be turned off.
873 ; This setting is global, rather than per-channel. It will not update on
876 ; Syntax is: cadence=ring,silence[,ring,silence[...]]
878 ; These are the default cadences:
880 ;cadence=125,125,2000,-4000
881 ;cadence=250,250,500,1000,250,250,500,-4000
882 ;cadence=125,125,125,125,125,-4000
883 ;cadence=1000,500,2500,-5000
885 ; Each channel consists of the channel number or range. It inherits the
886 ; parameters that were specified above its declaration.
889 ;callerid="Green Phone"<(256) 428-6121>
891 ;callerid="Black Phone"<(256) 428-6122>
893 ;callerid="CallerID Phone" <(630) 372-1564>
895 ;callerid="Pac Tel Phone" <(256) 428-6124>
897 ;callerid="Uniden Dead" <(256) 428-6125>
899 ;callerid="Cortelco 2500" <(256) 428-6126>
901 ;callerid="Main TA 750" <(256) 428-6127>
904 ; For example, maybe we have some other channels which start out in a
905 ; different context and use E & M signalling instead.
914 ; All those in group 0 I'll use for outgoing calls
916 ; Strip most significant digit (9) before sending
926 ;callerid="Joe Schmoe" <(256) 428-6131>
928 ;callerid="Megan May" <(256) 428-6132>
930 ;callerid="Suzy Queue" <(256) 428-6233>
932 ;callerid="Larry Moe" <(256) 428-6234>
935 ; Sample PRI (CPE) config: Specify the switchtype, the signalling as either
936 ; pri_cpe or pri_net for CPE or Network termination, and generally you will
937 ; want to create a single "group" for all channels of the PRI.
939 ; switchtype cannot be changed on a reload.
941 ; switchtype = national
942 ; signalling = pri_cpe
948 ; Used for distinctive ring support for x100p.
949 ; You can see the dringX patterns is to set any one of the dringXcontext fields
950 ; and they will be printed on the console when an inbound call comes in.
952 ; dringXrange is used to change the acceptable ranges for "tone offsets". Defaults to 10.
953 ; Note: a range of 0 is NOT what you might expect - it instead forces it to the default.
954 ; A range of -1 will force it to always match.
955 ; Anything lower than -1 would presumably cause it to never match.
958 ;dring1context=internal1
961 ;dring2context=internal2
963 ; If no pattern is matched here is where we go.
967 ; ---------------- Options for use with signalling=ss7 -----------------
968 ; None of them can be changed by a reload.
970 ; Variant of SS7 signalling:
971 ; Options are itu and ansi
974 ; SS7 Called Nature of Address Indicator
977 ; subscriber: Subscriber
979 ; international: International
980 ; dynamic: Dynamically selects the appropriate dialplan
982 ;ss7_called_nai=dynamic
984 ; SS7 Calling Nature of Address Indicator
987 ; subscriber: Subscriber
989 ; international: International
990 ; dynamic: Dynamically selects the appropriate dialplan
992 ;ss7_calling_nai=dynamic
995 ; sample 1 for Germany
996 ;ss7_internationalprefix = 00
997 ;ss7_nationalprefix = 0
998 ;ss7_subscriberprefix =
1002 ; This option is used to disable automatic sending of ACM when the call is started
1003 ; in the dialplan. If you do use this option, you will need to use the Proceeding()
1004 ; application in the dialplan to send ACM.
1007 ; All settings apply to linkset 1
1010 ; Point code of the linkset. For ITU, this is the decimal number
1011 ; format of the point code. For ANSI, this can either be in decimal
1012 ; number format or in the xxx-xxx-xxx format
1015 ; Point code of node adjacent to this signalling link (Possibly the STP between you and
1016 ; your destination). Point code format follows the same rules as above.
1019 ; Default point code that you would like to assign to outgoing messages (in case of
1020 ; routing through STPs, or using A links). Point code format follows the same rules
1024 ; Begin CIC (Circuit indication codes) count with this number
1027 ; What the MTP3 network indicator bits should be set to. Choices are
1028 ; national, national_spare, international, international_spare
1029 ;networkindicator=international
1031 ; First signalling channel
1034 ; Additional signalling channel for this linkset (So you can have a linkset
1035 ; with two signalling links in it). It seems like a silly way to do it, but
1036 ; for linksets with multiple signalling links, you add an additional sigchan
1037 ; line for every additional signalling link on the linkset.
1040 ; Channels to associate with CICs on this linkset
1043 ; For more information on setting up SS7, see the README file in libss7 or
1044 ; the doc/ss7.txt file in the Asterisk source tree.
1045 ; ----------------- SS7 Options ----------------------------------------
1047 ; ---------------- Options for use with signalling=mfcr2 --------------
1049 ; MFC-R2 signaling has lots of variants from country to country and even sometimes
1050 ; minor variants inside the same country. The only mandatory parameters here are:
1051 ; mfcr2_variant, mfcr2_max_ani and mfcr2_max_dnis.
1052 ; IT IS RECOMMENDED that you leave the default values (leaving it commented) for the
1053 ; other parameters unless you have problems or you have been instructed to change some
1054 ; parameter. OpenR2 library uses the mfcr2_variant parameter to try to determine the
1055 ; best defaults for your country, also refer to the OpenR2 package directory
1056 ; doc/asterisk/ where you can find sample configurations for some countries. If you
1057 ; want to contribute your configs for a particular country send them to the e-mail
1058 ; of the primary OpenR2 developer that you can find in the AUTHORS file of the OpenR2 package
1060 ; MFC/R2 variant. This depends on the OpenR2 supported variants
1061 ; A list of values can be found by executing the openr2 command r2test -l
1062 ; some valid values are:
1067 ; itu (per ITU spec)
1070 ; Max amount of ANI to ask for
1073 ; Max amount of DNIS to ask for
1076 ; whether or not to get the ANI before getting DNIS.
1077 ; some telcos require ANI first some others do not care
1078 ; if this go wrong, change this value
1079 ; mfcr2_get_ani_first=no
1081 ; Caller Category to send
1082 ; national_subscriber
1083 ; national_priority_subscriber
1084 ; international_subscriber
1085 ; international_priority_subscriber
1087 ; usually national_subscriber works just fine
1088 ; you can change this setting from the dialplan
1089 ; by setting the variable MFCR2_CATEGORY
1090 ; (remember to set _MFCR2_CATEGORY from originating channels)
1091 ; MFCR2_CATEGORY will also be a variable available in your context
1092 ; on incoming calls set to the value received from the far end
1093 ; mfcr2_category=national_subscriber
1095 ; Call logging is stored at the Asterisk
1096 ; logging directory specified in asterisk.conf
1097 ; plus mfcr2/<whatever you put here>
1098 ; if you specify 'span1' here and asterisk.conf has
1099 ; as logging directory /var/log/asterisk then the full
1100 ; path to your MFC/R2 call logs will be /var/log/asterisk/mfcr2/span1
1101 ; (the directory will be automatically created if not present already)
1102 ; remember to set mfcr2_call_files=yes
1103 ; mfcr2_logdir=span1
1105 ; whether or not to drop call files into mfcr2_logdir
1106 ; mfcr2_call_files=yes|no
1108 ; MFC/R2 valid logging values are: all,error,warning,debug,notice,cas,mf,stack,nothing
1109 ; error,warning,debug and notice are self-descriptive
1110 ; 'cas' is for logging ABCD CAS tx and rx
1111 ; 'mf' is for logging of the Multi Frequency tones
1112 ; 'stack' is for very verbose output of the channel and context call stack, only useful
1113 ; if you are debugging a crash or want to learn how the library works. The stack logging
1114 ; will be only enabled if the openr2 library was compiled with -DOR2_TRACE_STACKS
1115 ; You can mix up values, like: loglevel=error,debug,mf to log just error, debug and
1116 ; multi frequency messages
1117 ; 'all' is a special value to log all the activity
1118 ; 'nothing' is a clean-up value, in case you want to not log any activity for
1119 ; a channel or group of channels
1120 ; BE AWARE that the level of output logged will ALSO depend on
1121 ; the value you have in logger.conf, if you disable output in logger.conf
1122 ; then it does not matter you specify 'all' here, nothing will be logged
1123 ; so logger.conf has the last word on what is going to be logged
1126 ; MFC/R2 value in milliseconds for the MF timeout. Any negative value
1127 ; means 'default', smaller values than 500ms are not recommended
1128 ; and can cause malfunctioning. If you experience protocol error
1129 ; due to MF timeout try incrementing this value in 500ms steps
1130 ; mfcr2_mfback_timeout=-1
1132 ; MFC/R2 value in milliseconds for the metering pulse timeout.
1133 ; Metering pulses are sent by some telcos for some R2 variants
1134 ; during a call presumably for billing purposes to indicate costs,
1135 ; however this pulses use the same signal that is used to indicate
1136 ; call hangup, therefore a timeout is sometimes required to distinguish
1137 ; between a *real* hangup and a billing pulse that should not
1138 ; last more than 500ms, If you experience call drops after some
1139 ; minutes of being stablished try setting a value of some ms here,
1140 ; values greater than 500ms are not recommended.
1141 ; BE AWARE that choosing the proper protocol mfcr2_variant parameter
1142 ; implicitly sets a good recommended value for this timer, use this
1143 ; parameter only when you *really* want to override the default, otherwise
1144 ; just comment out this value or put a -1
1145 ; Any negative value means 'default'.
1146 ; mfcr2_metering_pulse_timeout=-1
1148 ; Brazil uses a special calling party category for collect calls (llamadas por cobrar)
1149 ; instead of using the operator (as in Mexico). The R2 spec in Brazil says a special GB tone
1150 ; should be used to reject collect calls. If you want to ALLOW collect calls specify 'yes',
1151 ; if you want to BLOCK collect calls then say 'no'. Default is to block collect calls.
1152 ; (see also 'mfcr2_double_answer')
1153 ; mfcr2_allow_collect_calls=no
1155 ; This feature is related but independent of mfcr2_allow_collect_calls
1156 ; Some PBX's require a double-answer process to block collect calls, if
1157 ; you ever have problems blocking collect calls using Group B signals (mfcr2_allow_collect_calls=no)
1158 ; then you may want to try with mfcr2_double_answer=yes, this will cause that every answer signal
1159 ; is changed by answer->clear back->answer (sort of a flash)
1160 ; (see also 'mfcr2_allow_collect_calls')
1161 ; mfcr2_double_answer=no
1163 ; This feature allows to skip the use of Group B/II signals and go directly
1164 ; to the accepted state for incoming calls
1165 ; mfcr2_immediate_accept=no
1167 ; You most likely dont need this feature. Default is yes.
1168 ; When this is set to yes, all calls that are offered (incoming calls) which
1169 ; DNIS is valid (exists in extensions.conf) and pass collect call validation
1170 ; will be accepted with a Group B tone (either call with charge or not, depending on mfcr2_charge_calls)
1171 ; with this set to 'no' then the call will NOT be accepted on offered, and the call will start its
1172 ; execution in extensions.conf without being accepted until the channel is answered (either with Answer() or
1173 ; any other application resulting in the channel being answered).
1174 ; This can be set to 'no' if your telco or PBX needs the hangup cause to be set accurately
1175 ; when this option is set to no you must explicitly accept the call with DAHDIAcceptR2Call
1176 ; or implicitly through the Answer() application.
1177 ; mfcr2_accept_on_offer=yes
1179 ; Skip request of calling party category and ANI
1180 ; you need openr2 >= 1.2.0 to use this feature
1181 ; mfcr2_skip_category=no
1183 ; WARNING: advanced users only! I really mean it
1184 ; this parameter is commented by default because
1185 ; YOU DON'T NEED IT UNLESS YOU REALLY GROK MFC/R2
1186 ; READ COMMENTS on doc/r2proto.conf in openr2 package
1188 ; mfcr2_advanced_protocol_file=/path/to/r2proto.conf
1190 ; Brazil use a special signal to force the release of the line (hangup) from the
1191 ; backward perspective. When mfcr2_forced_release=no, the normal clear back signal
1192 ; will be sent on hangup, which is OK for all mfcr2 variants I know of, except for
1193 ; Brazilian variant, where the central will leave the line up for several seconds (30, 60)
1194 ; which sometimes is not what people really want. When mfcr2_forced_release=yes, a different
1195 ; signal will be sent to hangup the call indicating that the line should be released immediately
1196 ; mfcr2_forced_release=no
1198 ; Whether or not report to the other end 'accept call with charge'
1199 ; This setting has no effect with most telecos, usually is safe
1200 ; leave the default (yes), but once in a while when interconnecting with
1201 ; old PBXs this may be useful.
1202 ; Concretely this affects the Group B signal used to accept calls
1203 ; The application DAHDIAcceptR2Call can also be used to decide this
1204 ; in the dial plan in a per-call basis instead of doing it here for all calls
1205 ; mfcr2_charge_calls=yes
1207 ; ---------------- END of options to be used with signalling=mfcr2
1209 ; Configuration Sections
1210 ; ~~~~~~~~~~~~~~~~~~~~~~
1211 ; You can also configure channels in a separate chan_dahdi.conf section. In
1212 ; this case the keyword 'channel' is not used. Instead the keyword
1213 ; 'dahdichan' is used (as in users.conf) - configuration is only processed
1214 ; in a section where the keyword dahdichan is used. It will only be
1215 ; processed in the end of the section. Thus the following section:
1222 ; Is somewhat equivalent to the following snippet in the section
1229 ; When starting a new section almost all of the configuration values are
1230 ; copied from their values at the end of the section [channels] in
1231 ; chan_dahdi.conf and [general] in users.conf - one section's configuration
1232 ; does not affect another one's.
1234 ; Instead of letting common configuration values "slide through" you can
1235 ; use configuration templates to easily keep the common part in one
1236 ; place and override where needed.
1243 ;threewaycalling = yes
1246 ;faxdetect = incoming
1250 ;callerid = My Name <501>
1251 ;mailbox = 501@mailboxes
projects / asterisk/asterisk.git / blob