2 ; Zapata telephony interface
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.
13 ; AUF : comme a priori la configuration d'interface ZAP sera toujours
14 ; locale, on fait uniquement un include
16 #include "auf/zapata.local"
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
25 ; Trunk groups are used for NFAS or GR-303 connections.
27 ; Group: Defines a trunk group.
28 ; trunkgroup => <trunkgroup>,<dchannel>[,<backup1>...]
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.
35 ;trunkgroup => 1,24,48
38 ; Spanmap: Associates a span with a trunk group
39 ; spanmap => <zapspan>,<trunkgroup>[,<logicalspan>]
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.
61 ; Switchtype: Only used for PRI.
63 ; national: National ISDN 2 (default)
64 ; dms100: Nortel DMS100
68 ; ni1: Old National ISDN 1
73 ; Some switches (AT&T especially) require network specific facility IE
74 ; supported values are currently 'none', 'sdn', 'megacom', 'tollfreemegacom', 'accunet'
78 ; PRI Dialplan: Only RARELY used for PRI.
81 ; private: Private ISDN
83 ; national: National ISDN
84 ; international: International ISDN
85 ; dynamic: Dynamically selects the appropriate dialplan
89 ; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's numbering plan)
92 ; private: Private ISDN
94 ; national: National ISDN
95 ; international: International ISDN
96 ; dynamic: Dynamically selects the appropriate dialplan
98 ;prilocaldialplan=national
100 ; PRI callerid prefixes based on the given TON/NPI (dialplan)
101 ; This is especially needed for euroisdn E1-PRIs
103 ; sample 1 for Germany
104 ;internationalprefix = 00
107 ;privateprefix = 07115678
110 ; sample 2 for Germany
111 ;internationalprefix = +
112 ;nationalprefix = +49
113 ;localprefix = +49711
114 ;privateprefix = +497115678
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*.
122 ;resetinterval = 3600
124 ; Overlap dialing mode (sending overlap digits)
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
133 ; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT
134 ; inband: Signal Busy/Congestion using in-band tones
136 ; priindication = outofband
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.
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)
156 ; pritimer => t200,1000
157 ; pritimer => t313,4000
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
164 ; Signalling method (default is fxs). Valid values:
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
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
191 ; The following are used for Radio interfaces:
192 ; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the
194 ; fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the
196 ; fxo_rx: Receive audio/COR on an FXO loopstart interface (FXS at the
198 ; fxo_tx: Transmit audio/PTT on an FXO groundstart interface (FXS at
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
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
209 ; sf_rxtx: Same as sf_txrx (for our dyslexic friends)
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.
220 ; outsignalling=featb
222 ; For Feature Group D Tandem access, to set the default CIC and OZZ use these
227 ; A variety of timing parameters can be specified as well
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)
238 ;rxwink=300 ; Atlas seems to use long (250ms) winks
240 ; How long generated tones (DTMF and MF) will be played on the channel
244 ; Whether or not to do distinctive ring detection on FXO lines
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.
250 ; Whether or not to use caller ID
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).
263 ; What signals the start of caller ID
264 ; ring = a ring signals the start
265 ; polarity = polarity reversal signals the start
269 ; Whether or not to hide outgoing caller ID (Override with *67 or *82)
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.
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
286 ; Whether or not use the caller ID presentation for the outgoing call that the
287 ; calling switch is sending.
288 ; See README.callingpres
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.
299 ; Support Caller*ID on Call Waiting
301 ;callwaitingcallerid=yes
303 ; Support three-way calling
307 ; Support flash-hook call transfer (requires three way calling)
308 ; Also enables call parking (overrides the 'canpark' parameter)
313 ; ('canpark=no' is overridden by 'transfer=yes')
317 ; Support call forward variable
321 ; Whether or not to support Call Return (*69)
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.
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.
333 ; for default voicemail context, the example below is fine:
337 ; for any other voicemail context, the following will produce the stutter tone:
339 ;mailbox=1234@context
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.
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.
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.
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.
361 ;echocancelwhenbridged=yes
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)
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
375 ; Note that these parameters do not apply to hardware echo cancellers.
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.
386 ; You may also set the default receive and transmit gains (in dB)
391 ; Logical groups can be assigned to allow outgoing rollover. Groups range
392 ; from 0 to 63, and multiple groups can be specified.
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.
405 ; Specify whether the channel should be answered immediately or if the simple
406 ; switch should provide dialtone, read digits, etc.
407 ; Note: If immediate=yes the dialplan execution will always start at extension
408 ; 's' priority 1 regardless of the dialed number!
412 ; Specify whether flash-hook transfers to 'busy' channels should complete or
413 ; return to the caller performing the transfer (default is yes).
417 ; CallerID can be set to "asreceived" or a specific number if you want to
418 ; override it. Note that "asreceived" only applies to trunk interfaces.
422 ; AMA flags affects the recording of Call Detail Records. If specified
423 ; it may be 'default', 'omit', 'billing', or 'documentation'.
427 ; Channels may be associated with an account code to ease
432 ; ADSI (Analog Display Services Interface) can be enabled on a per-channel
433 ; basis if you have (or may have) ADSI compatible CPE equipment
437 ; SMDI (Simplified Message Desk Interface) can be enabled on a per-channel
438 ; basis if you would like that channel to behave like an SMDI message desk.
439 ; The SMDI port specified should have already been defined in smdi.conf. The
440 ; default port is /dev/ttyS0.
445 ; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D
446 ; etc, it can be useful to perform busy detection either in an effort to
447 ; detect hangup or for detecting busies. This enables listening for
448 ; the beep-beep busy pattern.
452 ; If busydetect is enabled, it is also possible to specify how many busy tones
453 ; to wait for before hanging up. The default is 4, but better results can be
454 ; achieved if set to 6 or even 8. Mind that the higher the number, the more
455 ; time that will be needed to hangup a channel, but lowers the probability
456 ; that you will get random hangups.
460 ; If busydetect is enabled, it is also possible to specify the cadence of your
461 ; busy signal. In many countries, it is 500msec on, 500msec off. Without
462 ; busypattern specified, we'll accept any regular sound-silence pattern that
463 ; repeats <busycount> times as a busy signal. If you specify busypattern,
464 ; then we'll further check the length of the sound (tone) and silence, which
465 ; will further reduce the chance of a false positive.
469 ; NOTE: In the Asterisk Makefile you'll find further options to tweak the busy
470 ; detector. If your country has a busy tone with the same length tone and
471 ; silence (as many countries do), consider defining the
472 ; -DBUSYDETECT_COMPARE_TONE_AND_SILENCE option.
474 ; Use a polarity reversal to mark when a outgoing call is answered by the
477 ;answeronpolarityswitch=yes
479 ; In some countries, a polarity reversal is used to signal the disconnect of a
480 ; phone line. If the hanguponpolarityswitch option is selected, the call will
481 ; be considered "hung up" on a polarity reversal.
483 ;hanguponpolarityswitch=yes
485 ; On trunk interfaces (FXS) it can be useful to attempt to follow the progress
486 ; of a call through RINGING, BUSY, and ANSWERING. If turned on, call
487 ; progress attempts to determine answer, busy, and ringing on phone lines.
488 ; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers,
489 ; so don't count on it being very accurate.
491 ; Few zones are supported at the time of this writing, but may be selected
494 ; This feature can also easily detect false hangups. The symptoms of this is
495 ; being disconnected in the middle of a call for no reason.
500 ; FXO (FXS signalled) devices must have a timeout to determine if there was a
501 ; hangup before the line was answered. This value can be tweaked to shorten
502 ; how long it takes before Zap considers a non-ringing line to have hungup.
506 ; For FXO (FXS signalled) devices, whether to use pulse dial instead of DTMF
510 ; For fax detection, uncomment one of the following lines. The default is *OFF*
517 ; This option specifies a preference for which music on hold class this channel
518 ; should listen to when put on hold if the music class has not been set on the
519 ; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
520 ; channel putting this one on hold did not suggest a music class.
522 ; If this option is set to "passthrough", then the hold message will always be
523 ; passed through as signalling instead of generating hold music locally. This
524 ; setting is only valid when used on a channel that uses digital signalling.
526 ; This option may be specified globally, or on a per-user or per-peer basis.
528 ;mohinterpret=default
530 ; This option specifies which music on hold class to suggest to the peer channel
531 ; when this channel places the peer on hold. It may be specified globally or on
532 ; a per-user or per-peer basis.
536 ; PRI channels can have an idle extension and a minunused number. So long as
537 ; at least "minunused" channels are idle, chan_zap will try to call "idledial"
538 ; on them, and then dump them into the PBX in the "idleext" extension (which
539 ; is of the form exten@context). When channels are needed the "idle" calls
540 ; are disconnected (so long as there are at least "minidle" calls still
541 ; running, of course) to make more channels available. The primary use of
542 ; this is to create a dynamic service, where idle channels are bundled through
543 ; multilink PPP, thus more efficiently utilizing combined voice/data services
544 ; than conventional fixed mappings/muxings.
547 ;idleext=6999@dialout
551 ; Configure jitter buffers in zapata (each one is 20ms, default is 4)
555 ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
556 ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
557 ; ZAP channel. Defaults to "no". An enabled jitterbuffer will
558 ; be used only if the sending side can create and the receiving
559 ; side can not accept jitter. The ZAP channel can't accept jitter,
560 ; thus an enabled jitterbuffer on the receive ZAP side will always
561 ; be used if the sending side can create jitter.
563 ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
565 ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
566 ; resynchronized. Useful to improve the quality of the voice, with
567 ; big jumps in/broken timestamps, usually sent from exotic devices
568 ; and programs. Defaults to 1000.
570 ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a ZAP
571 ; channel. Two implementations are currently available - "fixed"
572 ; (with size always equals to jbmax-size) and "adaptive" (with
573 ; variable size, actually the new jb of IAX2). Defaults to fixed.
575 ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
576 ;-----------------------------------------------------------------------------------
578 ; You can define your own custom ring cadences here. You can define up to 8
579 ; pairs. If the silence is negative, it indicates where the callerid spill is
580 ; to be placed. Also, if you define any custom cadences, the default cadences
581 ; will be turned off.
583 ; Syntax is: cadence=ring,silence[,ring,silence[...]]
585 ; These are the default cadences:
587 ;cadence=125,125,2000,-4000
588 ;cadence=250,250,500,1000,250,250,500,-4000
589 ;cadence=125,125,125,125,125,-4000
590 ;cadence=1000,500,2500,-5000
592 ; Each channel consists of the channel number or range. It inherits the
593 ; parameters that were specified above its declaration.
595 ; For GR-303, CRV's are created like channels except they must start with the
596 ; trunk group followed by a colon, e.g.:
602 ;callerid="Green Phone"<(256) 428-6121>
604 ;callerid="Black Phone"<(256) 428-6122>
606 ;callerid="CallerID Phone" <(256) 428-6123>
607 ;callerid="CallerID Phone" <(630) 372-1564>
608 ;callerid="CallerID Phone" <(256) 704-4666>
610 ;callerid="Pac Tel Phone" <(256) 428-6124>
612 ;callerid="Uniden Dead" <(256) 428-6125>
614 ;callerid="Cortelco 2500" <(256) 428-6126>
616 ;callerid="Main TA 750" <(256) 428-6127>
619 ; For example, maybe we have some other channels which start out in a
620 ; different context and use E & M signalling instead.
629 ; All those in group 0 I'll use for outgoing calls
631 ; Strip most significant digit (9) before sending
641 ;callerid="Joe Schmoe" <(256) 428-6131>
643 ;callerid="Megan May" <(256) 428-6132>
645 ;callerid="Suzy Queue" <(256) 428-6233>
647 ;callerid="Larry Moe" <(256) 428-6234>
650 ; Sample PRI (CPE) config: Specify the switchtype, the signalling as either
651 ; pri_cpe or pri_net for CPE or Network termination, and generally you will
652 ; want to create a single "group" for all channels of the PRI.
654 ; switchtype = national
655 ; signalling = pri_cpe
661 ; Used for distinctive ring support for x100p.
662 ; You can see the dringX patterns is to set any one of the dringXcontext fields
663 ; and they will be printed on the console when an inbound call comes in.
666 ;dring1context=internal1
668 ;dring2context=internal2
669 ; If no pattern is matched here is where we go.