premier commit (version 1.4-24)
[asterisk-config-auf.git] / etc-asterisk / zapata.conf
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b2e905a6
TN
1;
2; Zapata telephony interface
3;
4; Configuration file
5;
6; You need to restart Asterisk to re-configure the Zap channel
7; CLI> reload chan_zap.so
8; will reload the configuration file,
9; but not all configuration options are
10; re-configured during a reload.
11
12
13; AUF : comme a priori la configuration d'interface ZAP sera toujours
14; locale, on fait uniquement un include
15
16#include "auf/zapata.local"
17
18; la suite est laissée comme exemple
19; NE PAS MODIFIER LE FICHIER sauf si vous savez ce que vous faites...
20; UTILISEZ PLUTOT /etc/asterisk/auf/zapata.local
21
22
23;[trunkgroups]
24;
25; Trunk groups are used for NFAS or GR-303 connections.
26;
27; Group: Defines a trunk group.
28; trunkgroup => <trunkgroup>,<dchannel>[,<backup1>...]
29;
30; trunkgroup is the numerical trunk group to create
31; dchannel is the zap channel which will have the
32; d-channel for the trunk.
33; backup1 is an optional list of backup d-channels.
34;
35;trunkgroup => 1,24,48
36;trunkgroup => 1,24
37;
38; Spanmap: Associates a span with a trunk group
39; spanmap => <zapspan>,<trunkgroup>[,<logicalspan>]
40;
41; zapspan is the zap span number to associate
42; trunkgroup is the trunkgroup (specified above) for the mapping
43; logicalspan is the logical span number within the trunk group to use.
44; if unspecified, no logical span number is used.
45;
46;spanmap => 1,1,1
47;spanmap => 2,1,2
48;spanmap => 3,1,3
49;spanmap => 4,1,4
50
51;[channels]
52;
53; Default language
54;
55;language=fr
56;
57; Default context
58;
59;context=default
60;
61; Switchtype: Only used for PRI.
62;
63; national: National ISDN 2 (default)
64; dms100: Nortel DMS100
65; 4ess: AT&T 4ESS
66; 5ess: Lucent 5ESS
67; euroisdn: EuroISDN
68; ni1: Old National ISDN 1
69; qsig: Q.SIG
70;
71;switchtype=national
72;
73; Some switches (AT&T especially) require network specific facility IE
74; supported values are currently 'none', 'sdn', 'megacom', 'tollfreemegacom', 'accunet'
75;
76;nsf=none
77;
78; PRI Dialplan: Only RARELY used for PRI.
79;
80; unknown: Unknown
81; private: Private ISDN
82; local: Local ISDN
83; national: National ISDN
84; international: International ISDN
85; dynamic: Dynamically selects the appropriate dialplan
86;
87;pridialplan=national
88;
89; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's numbering plan)
90;
91; unknown: Unknown
92; private: Private ISDN
93; local: Local ISDN
94; national: National ISDN
95; international: International ISDN
96; dynamic: Dynamically selects the appropriate dialplan
97;
98;prilocaldialplan=national
99;
100; PRI callerid prefixes based on the given TON/NPI (dialplan)
101; This is especially needed for euroisdn E1-PRIs
102;
103; sample 1 for Germany
104;internationalprefix = 00
105;nationalprefix = 0
106;localprefix = 0711
107;privateprefix = 07115678
108;unknownprefix =
109;
110; sample 2 for Germany
111;internationalprefix = +
112;nationalprefix = +49
113;localprefix = +49711
114;privateprefix = +497115678
115;unknownprefix =
116;
117; PRI resetinterval: sets the time in seconds between restart of unused
118; channels, defaults to 3600; minimum 60 seconds. Some PBXs don't like
119; channel restarts. so set the interval to a very long interval e.g. 100000000
120; or 'never' to disable *entirely*.
121;
122;resetinterval = 3600
123;
124; Overlap dialing mode (sending overlap digits)
125;
126;overlapdial=yes
127;
128; PRI Out of band indications.
129; Enable this to report Busy and Congestion on a PRI using out-of-band
130; notification. Inband indication, as used by Asterisk doesn't seem to work
131; with all telcos.
132;
133; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT
134; inband: Signal Busy/Congestion using in-band tones
135;
136; priindication = outofband
137;
138; If you need to override the existing channels selection routine and force all
139; PRI channels to be marked as exclusively selected, set this to yes.
140; priexclusive = yes
141;
142; ISDN Timers
143; All of the ISDN timers and counters that are used are configurable. Specify
144; the timer name, and its value (in ms for timers).
145; K: Layer 2 max number of outstanding unacknowledged I frames (default 7)
146; N200: Layer 2 max number of retransmissions of a frame (default 3)
147; T200: Layer 2 max time before retransmission of a frame (default 1000 ms)
148; T203: Layer 2 max time without frames being exchanged (default 10000 ms)
149; T305: Wait for DISCONNECT acknowledge (default 30000 ms)
150; T308: Wait for RELEASE acknowledge (default 4000 ms)
151; T309: Maintain active calls on Layer 2 disconnection (default -1, Asterisk clears calls)
152; EuroISDN: 6000 to 12000 ms, according to (N200 + 1) x T200 + 2s
153; May vary in other ISDN standards (Q.931 1993 : 90000 ms)
154; T313: Wait for CONNECT acknowledge, CPE side only (default 3000 ms)
155;
156; pritimer => t200,1000
157; pritimer => t313,4000
158;
159; To enable transmission of facility-based ISDN supplementary services (such
160; as caller name from CPE over facility), enable this option.
161; facilityenable = yes
162;
163;
164; Signalling method (default is fxs). Valid values:
165; em: E & M
166; em_w: E & M Wink
167; featd: Feature Group D (The fake, Adtran style, DTMF)
168; featdmf: Feature Group D (The real thing, MF (domestic, US))
169; featdmf_ta: Feature Group D (The real thing, MF (domestic, US)) through
170; a Tandem Access point
171; featb: Feature Group B (MF (domestic, US))
172; fgccama Feature Group C-CAMA (DP DNIS, MF ANI)
173; fgccamamf Feature Group C-CAMA MF (MF DNIS, MF ANI)
174; fxs_ls: FXS (Loop Start)
175; fxs_gs: FXS (Ground Start)
176; fxs_ks: FXS (Kewl Start)
177; fxo_ls: FXO (Loop Start)
178; fxo_gs: FXO (Ground Start)
179; fxo_ks: FXO (Kewl Start)
180; pri_cpe: PRI signalling, CPE side
181; pri_net: PRI signalling, Network side
182; gr303fxoks_net: GR-303 Signalling, FXO Loopstart, Network side
183; gr303fxsks_cpe: GR-303 Signalling, FXS Loopstart, CPE side
184; sf: SF (Inband Tone) Signalling
185; sf_w: SF Wink
186; sf_featd: SF Feature Group D (The fake, Adtran style, DTMF)
187; sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US))
188; sf_featb: SF Feature Group B (MF (domestic, US))
189; e911: E911 (MF) style signalling
190;
191; The following are used for Radio interfaces:
192; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the
193; channel bank)
194; fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the
195; channel bank)
196; fxo_rx: Receive audio/COR on an FXO loopstart interface (FXS at the
197; channel bank)
198; fxo_tx: Transmit audio/PTT on an FXO groundstart interface (FXS at
199; the channel bank)
200; em_rx: Receive audio/COR on an E&M interface (1-way)
201; em_tx: Transmit audio/PTT on an E&M interface (1-way)
202; em_txrx: Receive audio/COR AND Transmit audio/PTT on an E&M interface
203; (2-way)
204; em_rxtx: Same as em_txrx (for our dyslexic friends)
205; sf_rx: Receive audio/COR on an SF interface (1-way)
206; sf_tx: Transmit audio/PTT on an SF interface (1-way)
207; sf_txrx: Receive audio/COR AND Transmit audio/PTT on an SF interface
208; (2-way)
209; sf_rxtx: Same as sf_txrx (for our dyslexic friends)
210;
211;signalling=fxo_ls
212;
213; If you have an outbound signalling format that is different from format
214; specified above (but compatible), you can specify outbound signalling format,
215; (see below). The 'signalling' format specified will be the inbound signalling
216; format. If you only specify 'signalling', then it will be the format for
217; both inbound and outbound.
218;
219; signalling=featdmf
220; outsignalling=featb
221;
222; For Feature Group D Tandem access, to set the default CIC and OZZ use these
223; parameters:
224;defaultozz=0000
225;defaultcic=303
226;
227; A variety of timing parameters can be specified as well
228; Including:
229; prewink: Pre-wink time (default 50ms)
230; preflash: Pre-flash time (default 50ms)
231; wink: Wink time (default 150ms)
232; flash: Flash time (default 750ms)
233; start: Start time (default 1500ms)
234; rxwink: Receiver wink time (default 300ms)
235; rxflash: Receiver flashtime (default 1250ms)
236; debounce: Debounce timing (default 600ms)
237;
238;rxwink=300 ; Atlas seems to use long (250ms) winks
239;
240; How long generated tones (DTMF and MF) will be played on the channel
241; (in milliseconds)
242;toneduration=100
243;
244; Whether or not to do distinctive ring detection on FXO lines
245;
246;usedistinctiveringdetection=yes
247;distinctiveringaftercid=yes ; enable dring detection after callerid for those countries like Australia
248 ; where the ring cadence is changed *after* the callerid spill.
249;
250; Whether or not to use caller ID
251;
252;usecallerid=yes
253;
254; Type of caller ID signalling in use
255; bell = bell202 as used in US
256; v23 = v23 as used in the UK
257; v23_jp = v23 as used in Japan
258; dtmf = DTMF as used in Denmark, Sweden and Netherlands
259; smdi = Use SMDI for callerid. Requires SMDI to be enabled (usesmdi).
260;
261;cidsignalling=bell
262;
263; What signals the start of caller ID
264; ring = a ring signals the start
265; polarity = polarity reversal signals the start
266;
267;cidstart=ring
268;
269; Whether or not to hide outgoing caller ID (Override with *67 or *82)
270;
271;hidecallerid=no
272;
273; Whether or not to enable call waiting on internal extensions
274; With this set to 'yes', busy extensions will hear the call-waiting
275; tone, and can use hook-flash to switch between callers. The Dial()
276; app will not return the "BUSY" result for extensions.
277;
278;callwaiting=yes
279;
280; Whether or not restrict outgoing caller ID (will be sent as ANI only, not
281; available for the user)
282; Mostly use with FXS ports
283;
284;restrictcid=no
285;
286; Whether or not use the caller ID presentation for the outgoing call that the
287; calling switch is sending.
288; See README.callingpres
289;
290;usecallingpres=yes
291;
292; Some countries (UK) have ring tones with different ring tones (ring-ring),
293; which means the callerid needs to be set later on, and not just after
294; the first ring, as per the default.
295;
296;sendcalleridafter=1
297;
298;
299; Support Caller*ID on Call Waiting
300;
301;callwaitingcallerid=yes
302;
303; Support three-way calling
304;
305;threewaycalling=yes
306;
307; Support flash-hook call transfer (requires three way calling)
308; Also enables call parking (overrides the 'canpark' parameter)
309;
310;transfer=yes
311;
312; Allow call parking
313; ('canpark=no' is overridden by 'transfer=yes')
314;
315;canpark=yes
316;
317; Support call forward variable
318;
319;cancallforward=yes
320;
321; Whether or not to support Call Return (*69)
322;
323;callreturn=yes
324;
325; Stutter dialtone support: If a mailbox is specified without a voicemail
326; context, then when voicemail is received in a mailbox in the default
327; voicemail context in voicemail.conf, taking the phone off hook will cause a
328; stutter dialtone instead of a normal one.
329;
330; If a mailbox is specified *with* a voicemail context, the same will result
331; if voicemail received in mailbox in the specified voicemail context.
332;
333; for default voicemail context, the example below is fine:
334;
335;mailbox=1234
336;
337; for any other voicemail context, the following will produce the stutter tone:
338;
339;mailbox=1234@context
340;
341; Enable echo cancellation
342; Use either "yes", "no", or a power of two from 32 to 256 if you wish to
343; actually set the number of taps of cancellation.
344;
345; Note that when setting the number of taps, the number 256 does not translate
346; to 256 ms of echo cancellation. echocancel=256 means 256 / 8 = 32 ms.
347;
348; Note that if any of your Zaptel cards have hardware echo cancellers,
349; then this setting only turns them on and off; numeric settings will
350; be treated as "yes". There are no special settings required for
351; hardware echo cancellers; when present and enabled in their kernel
352; modules, they take precedence over the software echo canceller compiled
353; into Zaptel automatically.
354;
355;echocancel=yes
356;
357; Generally, it is not necessary (and in fact undesirable) to echo cancel when
358; the circuit path is entirely TDM. You may, however, change this behavior
359; by enabling the echo cancel during pure TDM bridging below.
360;
361;echocancelwhenbridged=yes
362;
363; In some cases, the echo canceller doesn't train quickly enough and there
364; is echo at the beginning of the call. Enabling echo training will cause
365; asterisk to briefly mute the channel, send an impulse, and use the impulse
366; response to pre-train the echo canceller so it can start out with a much
367; closer idea of the actual echo. Value may be "yes", "no", or a number of
368; milliseconds to delay before training (default = 400)
369;
370; WARNING: In some cases this option can make echo worse! If you are
371; trying to debug an echo problem, it is worth checking to see if your echo
372; is better with the option set to yes or no. Use whatever setting gives
373; the best results.
374;
375; Note that these parameters do not apply to hardware echo cancellers.
376;
377;echotraining=yes
378;echotraining=800
379;
380; If you are having trouble with DTMF detection, you can relax the DTMF
381; detection parameters. Relaxing them may make the DTMF detector more likely
382; to have "talkoff" where DTMF is detected when it shouldn't be.
383;
384;relaxdtmf=yes
385;
386; You may also set the default receive and transmit gains (in dB)
387;
388;rxgain=0.0
389;txgain=0.0
390;
391; Logical groups can be assigned to allow outgoing rollover. Groups range
392; from 0 to 63, and multiple groups can be specified.
393;
394;group=1
395;
396; Ring groups (a.k.a. call groups) and pickup groups. If a phone is ringing
397; and it is a member of a group which is one of your pickup groups, then
398; you can answer it by picking up and dialling *8#. For simple offices, just
399; make these both the same. Groups range from 0 to 63.
400;
401;callgroup=1
402;pickupgroup=1
403
404;
405; Specify whether the channel should be answered immediately or if the simple
406; switch should provide dialtone, read digits, etc.
407;
408;immediate=no
409;
410; Specify whether flash-hook transfers to 'busy' channels should complete or
411; return to the caller performing the transfer (default is yes).
412;
413;transfertobusy=no
414;
415; CallerID can be set to "asreceived" or a specific number if you want to
416; override it. Note that "asreceived" only applies to trunk interfaces.
417;
418;callerid=2564286000
419;
420; AMA flags affects the recording of Call Detail Records. If specified
421; it may be 'default', 'omit', 'billing', or 'documentation'.
422;
423;amaflags=default
424;
425; Channels may be associated with an account code to ease
426; billing
427;
428;accountcode=lss0101
429;
430; ADSI (Analog Display Services Interface) can be enabled on a per-channel
431; basis if you have (or may have) ADSI compatible CPE equipment
432;
433;adsi=yes
434;
435; SMDI (Simplified Message Desk Interface) can be enabled on a per-channel
436; basis if you would like that channel to behave like an SMDI message desk.
437; The SMDI port specified should have already been defined in smdi.conf. The
438; default port is /dev/ttyS0.
439;
440;usesmdi=yes
441;smdiport=/dev/ttyS0
442;
443; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D
444; etc, it can be useful to perform busy detection either in an effort to
445; detect hangup or for detecting busies. This enables listening for
446; the beep-beep busy pattern.
447;
448;busydetect=yes
449;
450; If busydetect is enabled, it is also possible to specify how many busy tones
451; to wait for before hanging up. The default is 4, but better results can be
452; achieved if set to 6 or even 8. Mind that the higher the number, the more
453; time that will be needed to hangup a channel, but lowers the probability
454; that you will get random hangups.
455;
456;busycount=4
457;
458; If busydetect is enabled, it is also possible to specify the cadence of your
459; busy signal. In many countries, it is 500msec on, 500msec off. Without
460; busypattern specified, we'll accept any regular sound-silence pattern that
461; repeats <busycount> times as a busy signal. If you specify busypattern,
462; then we'll further check the length of the sound (tone) and silence, which
463; will further reduce the chance of a false positive.
464;
465;busypattern=500,500
466;
467; NOTE: In the Asterisk Makefile you'll find further options to tweak the busy
468; detector. If your country has a busy tone with the same length tone and
469; silence (as many countries do), consider defining the
470; -DBUSYDETECT_COMPARE_TONE_AND_SILENCE option.
471;
472; Use a polarity reversal to mark when a outgoing call is answered by the
473; remote party.
474;
475;answeronpolarityswitch=yes
476;
477; In some countries, a polarity reversal is used to signal the disconnect of a
478; phone line. If the hanguponpolarityswitch option is selected, the call will
479; be considered "hung up" on a polarity reversal.
480;
481;hanguponpolarityswitch=yes
482;
483; On trunk interfaces (FXS) it can be useful to attempt to follow the progress
484; of a call through RINGING, BUSY, and ANSWERING. If turned on, call
485; progress attempts to determine answer, busy, and ringing on phone lines.
486; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers,
487; so don't count on it being very accurate.
488;
489; Few zones are supported at the time of this writing, but may be selected
490; with "progzone"
491;
492; This feature can also easily detect false hangups. The symptoms of this is
493; being disconnected in the middle of a call for no reason.
494;
495;callprogress=yes
496;progzone=us
497;
498; FXO (FXS signalled) devices must have a timeout to determine if there was a
499; hangup before the line was answered. This value can be tweaked to shorten
500; how long it takes before Zap considers a non-ringing line to have hungup.
501;
502;ringtimeout=8000
503;
504; For FXO (FXS signalled) devices, whether to use pulse dial instead of DTMF
505;
506;pulsedial=yes
507;
508; For fax detection, uncomment one of the following lines. The default is *OFF*
509;
510;faxdetect=both
511;faxdetect=incoming
512;faxdetect=outgoing
513;faxdetect=no
514;
515; This option specifies a preference for which music on hold class this channel
516; should listen to when put on hold if the music class has not been set on the
517; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
518; channel putting this one on hold did not suggest a music class.
519;
520; If this option is set to "passthrough", then the hold message will always be
521; passed through as signalling instead of generating hold music locally. This
522; setting is only valid when used on a channel that uses digital signalling.
523;
524; This option may be specified globally, or on a per-user or per-peer basis.
525;
526;mohinterpret=default
527;
528; This option specifies which music on hold class to suggest to the peer channel
529; when this channel places the peer on hold. It may be specified globally or on
530; a per-user or per-peer basis.
531;
532;mohsuggest=default
533;
534; PRI channels can have an idle extension and a minunused number. So long as
535; at least "minunused" channels are idle, chan_zap will try to call "idledial"
536; on them, and then dump them into the PBX in the "idleext" extension (which
537; is of the form exten@context). When channels are needed the "idle" calls
538; are disconnected (so long as there are at least "minidle" calls still
539; running, of course) to make more channels available. The primary use of
540; this is to create a dynamic service, where idle channels are bundled through
541; multilink PPP, thus more efficiently utilizing combined voice/data services
542; than conventional fixed mappings/muxings.
543;
544;idledial=6999
545;idleext=6999@dialout
546;minunused=2
547;minidle=1
548;
549; Configure jitter buffers in zapata (each one is 20ms, default is 4)
550;
551;jitterbuffers=4
552;
553;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
554; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
555 ; ZAP channel. Defaults to "no". An enabled jitterbuffer will
556 ; be used only if the sending side can create and the receiving
557 ; side can not accept jitter. The ZAP channel can't accept jitter,
558 ; thus an enabled jitterbuffer on the receive ZAP side will always
559 ; be used if the sending side can create jitter.
560
561; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
562
563; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
564 ; resynchronized. Useful to improve the quality of the voice, with
565 ; big jumps in/broken timestamps, usually sent from exotic devices
566 ; and programs. Defaults to 1000.
567
568; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a ZAP
569 ; channel. Two implementations are currently available - "fixed"
570 ; (with size always equals to jbmax-size) and "adaptive" (with
571 ; variable size, actually the new jb of IAX2). Defaults to fixed.
572
573; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
574;-----------------------------------------------------------------------------------
575;
576; You can define your own custom ring cadences here. You can define up to 8
577; pairs. If the silence is negative, it indicates where the callerid spill is
578; to be placed. Also, if you define any custom cadences, the default cadences
579; will be turned off.
580;
581; Syntax is: cadence=ring,silence[,ring,silence[...]]
582;
583; These are the default cadences:
584;
585;cadence=125,125,2000,-4000
586;cadence=250,250,500,1000,250,250,500,-4000
587;cadence=125,125,125,125,125,-4000
588;cadence=1000,500,2500,-5000
589;
590; Each channel consists of the channel number or range. It inherits the
591; parameters that were specified above its declaration.
592;
593; For GR-303, CRV's are created like channels except they must start with the
594; trunk group followed by a colon, e.g.:
595;
596; crv => 1:1
597; crv => 2:1-2,5-8
598;
599;
600;callerid="Green Phone"<(256) 428-6121>
601;channel => 1
602;callerid="Black Phone"<(256) 428-6122>
603;channel => 2
604;callerid="CallerID Phone" <(256) 428-6123>
605;callerid="CallerID Phone" <(630) 372-1564>
606;callerid="CallerID Phone" <(256) 704-4666>
607;channel => 3
608;callerid="Pac Tel Phone" <(256) 428-6124>
609;channel => 4
610;callerid="Uniden Dead" <(256) 428-6125>
611;channel => 5
612;callerid="Cortelco 2500" <(256) 428-6126>
613;channel => 6
614;callerid="Main TA 750" <(256) 428-6127>
615;channel => 44
616;
617; For example, maybe we have some other channels which start out in a
618; different context and use E & M signalling instead.
619;
620;context=remote
621;sigalling=em
622;channel => 15
623;channel => 16
624
625;signalling=em_w
626;
627; All those in group 0 I'll use for outgoing calls
628;
629; Strip most significant digit (9) before sending
630;
631;stripmsd=1
632;callerid=asreceived
633;group=0
634;signalling=fxs_ls
635;channel => 45
636
637;signalling=fxo_ls
638;group=1
639;callerid="Joe Schmoe" <(256) 428-6131>
640;channel => 25
641;callerid="Megan May" <(256) 428-6132>
642;channel => 26
643;callerid="Suzy Queue" <(256) 428-6233>
644;channel => 27
645;callerid="Larry Moe" <(256) 428-6234>
646;channel => 28
647;
648; Sample PRI (CPE) config: Specify the switchtype, the signalling as either
649; pri_cpe or pri_net for CPE or Network termination, and generally you will
650; want to create a single "group" for all channels of the PRI.
651;
652; switchtype = national
653; signalling = pri_cpe
654; group = 2
655; channel => 1-23
656
657;
658
659; Used for distinctive ring support for x100p.
660; You can see the dringX patterns is to set any one of the dringXcontext fields
661; and they will be printed on the console when an inbound call comes in.
662;
663;dring1=95,0,0
664;dring1context=internal1
665;dring2=325,95,0
666;dring2context=internal2
667; If no pattern is matched here is where we go.
668;context=default
669;channel => 1
670