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 (signalling, as well as
11 ; PRI and SS7-related settings cannot be changed on a
14 ; This file documents many configuration variables. Normally unless you
15 ; know what a variable means or that it should be changed, there's no
16 ; reason to unrem lines.
18 ; remmed-out examples below (those lines that begin with a ';' but no
19 ; space afterwards) typically show a value that is not the defauult value,
20 ; but would make sense under cetain circumstances. The default values
21 ; are usually sane. Thus you should typically not touch them unless you
22 ; know what they mean or you know you should change them.
27 ; Trunk groups are used for NFAS or GR-303 connections.
29 ; Group: Defines a trunk group.
30 ; trunkgroup => <trunkgroup>,<dchannel>[,<backup1>...]
32 ; trunkgroup is the numerical trunk group to create
33 ; dchannel is the zap channel which will have the
34 ; d-channel for the trunk.
35 ; backup1 is an optional list of backup d-channels.
37 ;trunkgroup => 1,24,48
40 ; Spanmap: Associates a span with a trunk group
41 ; spanmap => <zapspan>,<trunkgroup>[,<logicalspan>]
43 ; zapspan is the zap span number to associate
44 ; trunkgroup is the trunkgroup (specified above) for the mapping
45 ; logicalspan is the logical span number within the trunk group to use.
46 ; if unspecified, no logical span number is used.
59 ; Context for calls. Defaults to 'default'
63 ; Switchtype: Only used for PRI.
65 ; national: National ISDN 2 (default)
66 ; dms100: Nortel DMS100
69 ; euroisdn: EuroISDN (common in Europe)
70 ; ni1: Old National ISDN 1
75 ; Some switches (AT&T especially) require network specific facility IE
76 ; supported values are currently 'none', 'sdn', 'megacom', 'tollfreemegacom', 'accunet'
78 ; nsf cannot be changed on a reload.
82 ; PRI Dialplan: The ISDN-level Type Of Number (TON) or numbering plan, used for
83 ; the dialed number. For most installations, leaving this as 'unknown' (the
84 ; default) works in the most cases. In some very unusual circumstances, you
85 ; may need to set this to 'dynamic' or 'redundant'. Note that if you set one
86 ; of the others, you will be unable to dial another class of numbers. For
87 ; example, if you set 'national', you will be unable to dial local or
88 ; international numbers.
90 ; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's
91 ; numbering plan). In North America, the typical use is sending the 10 digit
92 ; callerID number and setting the prilocaldialplan to 'national' (the default).
93 ; Only VERY rarely will you need to change this.
95 ; Neither pridialplan nor prilocaldialplan can be changed on reload.
98 ; private: Private ISDN
100 ; national: National ISDN
101 ; international: International ISDN
102 ; dynamic: Dynamically selects the appropriate dialplan
103 ; redundant: Same as dynamic, except that the underlying number is not
104 ; changed (not common)
107 ;prilocaldialplan=national
109 ; pridialplan may be also set at dialtime, by prefixing the dialled number with
110 ; one of the following letters:
114 ; L - Local (Net Specific)
117 ; R - Reserved (should probably never be used but is included for completeness)
119 ; Additionally, you may also set the following NPI bits (also by prefixing the
120 ; dialled string with one of the following letters):
122 ; e - E.163/E.164 (ISDN/telephony)
127 ; r - Reserved (should probably never be used but is included for completeness)
129 ; You may also set the prilocaldialplan in the same way, but by prefixing the
130 ; Caller*ID Number, rather than the dialled number. Please note that telcos
131 ; which require this kind of additional manipulation of the TON/NPI are *rare*.
132 ; Most telco PRIs will work fine simply by setting pridialplan to unknown or
136 ; PRI caller ID prefixes based on the given TON/NPI (dialplan)
137 ; This is especially needed for EuroISDN E1-PRIs
139 ; None of the prefix settings can be changed on reload.
141 ; sample 1 for Germany
142 ;internationalprefix = 00
145 ;privateprefix = 07115678
148 ; sample 2 for Germany
149 ;internationalprefix = +
150 ;nationalprefix = +49
151 ;localprefix = +49711
152 ;privateprefix = +497115678
155 ; PRI resetinterval: sets the time in seconds between restart of unused
156 ; B channels; defaults to 'never'.
158 ;resetinterval = 3600
160 ; Overlap dialing mode (sending overlap digits)
161 ; Cannot be changed on a reload.
165 ; PRI Out of band indications.
166 ; Enable this to report Busy and Congestion on a PRI using out-of-band
167 ; notification. Inband indication, as used by Asterisk doesn't seem to work
170 ; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT
171 ; inband: Signal Busy/Congestion using in-band tones (default)
173 ; priindication cannot be changed on a reload.
175 ;priindication = outofband
177 ; If you need to override the existing channels selection routine and force all
178 ; PRI channels to be marked as exclusively selected, set this to yes.
180 ; priexclusive cannot be changed on a reload.
185 ; All of the ISDN timers and counters that are used are configurable. Specify
186 ; the timer name, and its value (in ms for timers).
187 ; K: Layer 2 max number of outstanding unacknowledged I frames (default 7)
188 ; N200: Layer 2 max number of retransmissions of a frame (default 3)
189 ; T200: Layer 2 max time before retransmission of a frame (default 1000 ms)
190 ; T203: Layer 2 max time without frames being exchanged (default 10000 ms)
191 ; T305: Wait for DISCONNECT acknowledge (default 30000 ms)
192 ; T308: Wait for RELEASE acknowledge (default 4000 ms)
193 ; T309: Maintain active calls on Layer 2 disconnection (default -1,
194 ; Asterisk clears calls)
195 ; EuroISDN: 6000 to 12000 ms, according to (N200 + 1) x T200 + 2s
196 ; May vary in other ISDN standards (Q.931 1993 : 90000 ms)
197 ; T313: Wait for CONNECT acknowledge, CPE side only (default 3000 ms)
199 ;pritimer => t200,1000
200 ;pritimer => t313,4000
202 ; To enable transmission of facility-based ISDN supplementary services (such
203 ; as caller name from CPE over facility), enable this option.
204 ; Cannot be changed on a reload.
206 ;facilityenable = yes
208 ; pritimer cannot be changed on a reload.
210 ; Signalling method. The default is "auto". Valid values:
211 ; auto: Use the current value from Zaptel.
215 ; featd: Feature Group D (The fake, Adtran style, DTMF)
216 ; featdmf: Feature Group D (The real thing, MF (domestic, US))
217 ; featdmf_ta: Feature Group D (The real thing, MF (domestic, US)) through
218 ; a Tandem Access point
219 ; featb: Feature Group B (MF (domestic, US))
220 ; fgccama Feature Group C-CAMA (DP DNIS, MF ANI)
221 ; fgccamamf Feature Group C-CAMA MF (MF DNIS, MF ANI)
222 ; fxs_ls: FXS (Loop Start)
223 ; fxs_gs: FXS (Ground Start)
224 ; fxs_ks: FXS (Kewl Start)
225 ; fxo_ls: FXO (Loop Start)
226 ; fxo_gs: FXO (Ground Start)
227 ; fxo_ks: FXO (Kewl Start)
228 ; pri_cpe: PRI signalling, CPE side
229 ; pri_net: PRI signalling, Network side
230 ; gr303fxoks_net: GR-303 Signalling, FXO Loopstart, Network side
231 ; gr303fxsks_cpe: GR-303 Signalling, FXS Loopstart, CPE side
232 ; sf: SF (Inband Tone) Signalling
234 ; sf_featd: SF Feature Group D (The fake, Adtran style, DTMF)
235 ; sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US))
236 ; sf_featb: SF Feature Group B (MF (domestic, US))
237 ; e911: E911 (MF) style signalling
238 ; ss7: Signalling System 7
240 ; The following are used for Radio interfaces:
241 ; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the
243 ; fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the
245 ; fxo_rx: Receive audio/COR on an FXO loopstart interface (FXS at the
247 ; fxo_tx: Transmit audio/PTT on an FXO groundstart interface (FXS at
249 ; em_rx: Receive audio/COR on an E&M interface (1-way)
250 ; em_tx: Transmit audio/PTT on an E&M interface (1-way)
251 ; em_txrx: Receive audio/COR AND Transmit audio/PTT on an E&M interface
253 ; em_rxtx: Same as em_txrx (for our dyslexic friends)
254 ; sf_rx: Receive audio/COR on an SF interface (1-way)
255 ; sf_tx: Transmit audio/PTT on an SF interface (1-way)
256 ; sf_txrx: Receive audio/COR AND Transmit audio/PTT on an SF interface
258 ; sf_rxtx: Same as sf_txrx (for our dyslexic friends)
259 ; ss7: Signalling System 7
261 ; signalling of a channel can not be changed on a reload.
265 ; If you have an outbound signalling format that is different from format
266 ; specified above (but compatible), you can specify outbound signalling format,
267 ; (see below). The 'signalling' format specified will be the inbound signalling
268 ; format. If you only specify 'signalling', then it will be the format for
269 ; both inbound and outbound.
271 ; outsignalling can only be one of:
272 ; em, em_e1, em_w, sf, sf_w, sf_featd, sf_featdmf, sf_featb, featd,
273 ; featdmf, featdmf_ta, e911, fgccama, fgccamamf
275 ; outsignalling cannot be changed on a reload.
281 ; For Feature Group D Tandem access, to set the default CIC and OZZ use these
282 ; parameters (Will not be updated on reload):
287 ; A variety of timing parameters can be specified as well
288 ; The default values for those are "-1", which is to use the
289 ; compile-time defaults of the Zaptel kernel modules. The timing
290 ; parameters, (with the standard default from Zaptel):
292 ; prewink: Pre-wink time (default 50ms)
293 ; preflash: Pre-flash time (default 50ms)
294 ; wink: Wink time (default 150ms)
295 ; flash: Flash time (default 750ms)
296 ; start: Start time (default 1500ms)
297 ; rxwink: Receiver wink time (default 300ms)
298 ; rxflash: Receiver flashtime (default 1250ms)
299 ; debounce: Debounce timing (default 600ms)
301 ; None of them will update on a reload.
303 ; How long generated tones (DTMF and MF) will be played on the channel
306 ; This is a global, rather than a per-channel setting. It will not be
307 ; updated on a reload.
311 ; Whether or not to do distinctive ring detection on FXO lines:
313 ;usedistinctiveringdetection=yes
315 ; enable dring detection after caller ID for those countries like Australia
316 ; where the ring cadence is changed *after* the caller ID spill:
318 ;distinctiveringaftercid=yes
320 ; Whether or not to use caller ID:
324 ; Hide the name part and leave just the number part of the caller ID
325 ; string. Only applies to PRI channels.
326 ;hidecalleridname=yes
328 ; Type of caller ID signalling in use
329 ; bell = bell202 as used in US (default)
330 ; v23 = v23 as used in the UK
331 ; v23_jp = v23 as used in Japan
332 ; dtmf = DTMF as used in Denmark, Sweden and Netherlands
333 ; smdi = Use SMDI for caller ID. Requires SMDI to be enabled (usesmdi).
337 ; What signals the start of caller ID
338 ; ring = a ring signals the start (default)
339 ; polarity = polarity reversal signals the start
340 ; polarity_IN = polarity reversal signals the start, for India,
341 ; for dtmf dialtone detection; using DTMF.
342 ; (see doc/India-CID.txt)
346 ; Whether or not to hide outgoing caller ID (Override with *67 or *82)
347 ; (If your dialplan doesn't catch it)
351 ; The following option enables receiving MWI on FXO lines. The default
352 ; value is no. When this is enabled, and MWI notification indicates on or off,
353 ; the script specified by the mwimonitornotify option is executed. Also, an
354 ; internal Asterisk MWI event will be generated so that any other part of
355 ; Asterisk that cares about MWI state changes will get notified, just as if
356 ; the state change came from app_voicemail. The energy level that must be seen
357 ; before starting the MWI detection process can be set with 'mwilevel'.
362 ; This option is used in conjunction with mwimonitor. This will get executed
363 ; when incoming MWI state changes. The script is passed 2 arguments. The
364 ; first is the corresponding mailbox, and the second is 1 or 0, indicating if
365 ; there are messages waiting or not.
367 ;mwimonitornotify=/usr/local/bin/zapnotify.sh
369 ; Whether or not to enable call waiting on internal extensions
370 ; With this set to 'yes', busy extensions will hear the call-waiting
371 ; tone, and can use hook-flash to switch between callers. The Dial()
372 ; app will not return the "BUSY" result for extensions.
376 ; Whether or not restrict outgoing caller ID (will be sent as ANI only, not
377 ; available for the user)
378 ; Mostly use with FXS ports
382 ; Whether or not use the caller ID presentation for the outgoing call that the
383 ; calling switch is sending.
384 ; See README.callingpres. FIXME: file no longer exists.
388 ; Some countries (UK) have ring tones with different ring tones (ring-ring),
389 ; which means the caller ID needs to be set later on, and not just after
390 ; the first ring, as per the default (1).
392 ;sendcalleridafter = 2
395 ; Support caller ID on Call Waiting
397 callwaitingcallerid=yes
399 ; Support three-way calling
403 ; For FXS ports (either direct analog or over T1/E1):
404 ; Support flash-hook call transfer (requires three way calling)
405 ; Also enables call parking (overrides the 'canpark' parameter)
407 ; For digital ports using ISDN PRI protocols:
408 ; Support switch-side transfer (called 2BCT, RLT or other names)
409 ; This setting must be enabled on both ports involved, and the
410 ; 'facilityenable' setting must also be enabled to allow sending
411 ; the transfer to the ISDN switch, since it sent in a FACILITY
417 ; ('canpark=no' is overridden by 'transfer=yes')
421 ; Support call forward variable
425 ; Whether or not to support Call Return (*69, if your dialplan doesn't
430 ; Stutter dialtone support: If a mailbox is specified without a voicemail
431 ; context, then when voicemail is received in a mailbox in the default
432 ; voicemail context in voicemail.conf, taking the phone off hook will cause a
433 ; stutter dialtone instead of a normal one.
435 ; If a mailbox is specified *with* a voicemail context, the same will result
436 ; if voicemail received in mailbox in the specified voicemail context.
438 ; for default voicemail context, the example below is fine:
442 ; for any other voicemail context, the following will produce the stutter tone:
444 ;mailbox=1234@context
446 ; Enable echo cancellation
447 ; Use either "yes", "no", or a power of two from 32 to 256 if you wish to
448 ; actually set the number of taps of cancellation.
450 ; Note that when setting the number of taps, the number 256 does not translate
451 ; to 256 ms of echo cancellation. echocancel=256 means 256 / 8 = 32 ms.
453 ; Note that if any of your Zaptel cards have hardware echo cancellers,
454 ; then this setting only turns them on and off; numeric settings will
455 ; be treated as "yes". There are no special settings required for
456 ; hardware echo cancellers; when present and enabled in their kernel
457 ; modules, they take precedence over the software echo canceller compiled
458 ; into Zaptel automatically.
463 ; As of Zaptel 1.4.8, some Zaptel echo cancellers (software and hardware)
464 ; support adjustable parameters; these parameters can be supplied as
465 ; additional options to the 'echocancel' setting. Note that Asterisk
466 ; does not attempt to validate the parameters or their values, so if you
467 ; supply an invalid parameter you will not know the specific reason it
468 ; failed without checking the kernel message log for the error(s)
469 ; put there by Zaptel.
471 ;echocancel=128,param1=32,param2=0,param3=14
473 ; Generally, it is not necessary (and in fact undesirable) to echo cancel when
474 ; the circuit path is entirely TDM. You may, however, change this behavior
475 ; by enabling the echo canceller during pure TDM bridging below.
477 echocancelwhenbridged=yes
479 ; In some cases, the echo canceller doesn't train quickly enough and there
480 ; is echo at the beginning of the call. Enabling echo training will cause
481 ; Zaptel to briefly mute the channel, send an impulse, and use the impulse
482 ; response to pre-train the echo canceller so it can start out with a much
483 ; closer idea of the actual echo. Value may be "yes", "no", or a number of
484 ; milliseconds to delay before training (default = 400)
486 ; WARNING: In some cases this option can make echo worse! If you are
487 ; trying to debug an echo problem, it is worth checking to see if your echo
488 ; is better with the option set to yes or no. Use whatever setting gives
491 ; Note that these parameters do not apply to hardware echo cancellers.
496 ; If you are having trouble with DTMF detection, you can relax the DTMF
497 ; detection parameters. Relaxing them may make the DTMF detector more likely
498 ; to have "talkoff" where DTMF is detected when it shouldn't be.
502 ; You may also set the default receive and transmit gains (in dB)
504 ; Gain Settings: increasing / decreasing the volume level on a channel.
505 ; The values are in db (decibells). A positive number
506 ; increases the volume level on a channel, and a
507 ; negavive value decreases volume level.
509 ; There are several independent gain settings:
510 ; rxgain: gain for the rx (receive - into Asterisk) channel. Default: 0.0
511 ; txgain: gain for the tx (transmit - out of Asterisk Asterisk) channel.
513 ; cid_rxgain: set the gain just for the caller ID sounds Asterisk
514 ; emits. Default: 5.0 .
519 ; Logical groups can be assigned to allow outgoing roll-over. Groups range
520 ; from 0 to 63, and multiple groups can be specified. By default the
521 ; channel is not a member of any group.
523 ; Note that an explicit empty value for 'group' is invalid, and will not
524 ; override a previous non-empty one. The same applies to callgroup and
525 ; pickupgroup as well.
529 ; Ring groups (a.k.a. call groups) and pickup groups. If a phone is ringing
530 ; and it is a member of a group which is one of your pickup groups, then
531 ; you can answer it by picking up and dialing *8#. For simple offices, just
532 ; make these both the same. Groups range from 0 to 63.
537 ; Channel variable to be set for all calls from this channel
541 ; Specify whether the channel should be answered immediately or if the simple
542 ; switch should provide dialtone, read digits, etc.
543 ; Note: If immediate=yes the dialplan execution will always start at extension
544 ; 's' priority 1 regardless of the dialed number!
548 ; Specify whether flash-hook transfers to 'busy' channels should complete or
549 ; return to the caller performing the transfer (default is yes).
553 ; caller ID can be set to "asreceived" or a specific number if you want to
554 ; override it. Note that "asreceived" only applies to trunk interfaces.
555 ; fullname sets just the
557 ; fullname: sets just the name part.
558 ; cid_number: sets just the number part:
562 ;callerid = My Name <2564286000>
563 ; Which can also be written as:
564 ;cid_number = 2564286000
567 ;callerid = asreceived
569 ; should we use the caller ID from incoming call on zap transfer?
571 ;useincomingcalleridonzaptransfer = yes
573 ; AMA flags affects the recording of Call Detail Records. If specified
574 ; it may be 'default', 'omit', 'billing', or 'documentation'.
578 ; Channels may be associated with an account code to ease
583 ; ADSI (Analog Display Services Interface) can be enabled on a per-channel
584 ; basis if you have (or may have) ADSI compatible CPE equipment
588 ; SMDI (Simplified Message Desk Interface) can be enabled on a per-channel
589 ; basis if you would like that channel to behave like an SMDI message desk.
590 ; The SMDI port specified should have already been defined in smdi.conf. The
591 ; default port is /dev/ttyS0.
596 ; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D
597 ; etc, it can be useful to perform busy detection either in an effort to
598 ; detect hangup or for detecting busies. This enables listening for
599 ; the beep-beep busy pattern.
603 ; If busydetect is enabled, it is also possible to specify how many busy tones
604 ; to wait for before hanging up. The default is 3, but it might be
605 ; safer to set to 6 or even 8. Mind that the higher the number, the more
606 ; time that will be needed to hangup a channel, but lowers the probability
607 ; that you will get random hangups.
611 ; If busydetect is enabled, it is also possible to specify the cadence of your
612 ; busy signal. In many countries, it is 500msec on, 500msec off. Without
613 ; busypattern specified, we'll accept any regular sound-silence pattern that
614 ; repeats <busycount> times as a busy signal. If you specify busypattern,
615 ; then we'll further check the length of the sound (tone) and silence, which
616 ; will further reduce the chance of a false positive.
620 ; NOTE: In make menuselect, you'll find further options to tweak the busy
621 ; detector. If your country has a busy tone with the same length tone and
622 ; silence (as many countries do), consider enabling the
623 ; BUSYDETECT_COMPARE_TONE_AND_SILENCE option.
625 ; To further detect which hangup tone your telco provider is sending, it is
626 ; useful to use the ztmonitor utility to record the audio that main/dsp.c
627 ; is receiving after the caller hangs up.
629 ; Use a polarity reversal to mark when a outgoing call is answered by the
632 ;answeronpolarityswitch=yes
634 ; In some countries, a polarity reversal is used to signal the disconnect of a
635 ; phone line. If the hanguponpolarityswitch option is selected, the call will
636 ; be considered "hung up" on a polarity reversal.
638 ;hanguponpolarityswitch=yes
640 ; polarityonanswerdelay: minimal time period (ms) between the answer
641 ; polarity switch and hangup polarity switch.
644 ; On trunk interfaces (FXS) it can be useful to attempt to follow the progress
645 ; of a call through RINGING, BUSY, and ANSWERING. If turned on, call
646 ; progress attempts to determine answer, busy, and ringing on phone lines.
647 ; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers,
648 ; so don't count on it being very accurate.
650 ; Few zones are supported at the time of this writing, but may be selected
653 ; progzone also affects the pattern used for buzydetect (unless
654 ; busypattern is set explicitly). The possible values are:
656 ; ca (alias for 'us')
658 ; br (Brazil, alias for 'cr')
661 ; This feature can also easily detect false hangups. The symptoms of this is
662 ; being disconnected in the middle of a call for no reason.
667 ; Set the tonezone. Equivalent of the defaultzone settings in
668 ; /etc/zaptel.conf . This sets the tone zone by number.
669 ; Note that you'd still need to load tonezones (loadzone in zaptel.conf).
670 ; The default is -1: not to set anything.
671 ;tonezone = 0 ; 0 is US
673 ; FXO (FXS signalled) devices must have a timeout to determine if there was a
674 ; hangup before the line was answered. This value can be tweaked to shorten
675 ; how long it takes before Zap considers a non-ringing line to have hungup.
677 ; ringtimeout will not update on a reload.
681 ; For FXO (FXS signalled) devices, whether to use pulse dial instead of DTMF
682 ; Pulse digits from phones (FXS devices, FXO signalling) are always
687 ; For fax detection, uncomment one of the following lines. The default is *OFF*
694 ; This option specifies a preference for which music on hold class this channel
695 ; should listen to when put on hold if the music class has not been set on the
696 ; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
697 ; channel putting this one on hold did not suggest a music class.
699 ; If this option is set to "passthrough", then the hold message will always be
700 ; passed through as signalling instead of generating hold music locally. This
701 ; setting is only valid when used on a channel that uses digital signalling.
703 ;mohinterpret=default
705 ; This option specifies which music on hold class to suggest to the peer channel
706 ; when this channel places the peer on hold.
710 ; PRI channels can have an idle extension and a minunused number. So long as
711 ; at least "minunused" channels are idle, chan_zap will try to call "idledial"
712 ; on them, and then dump them into the PBX in the "idleext" extension (which
713 ; is of the form exten@context). When channels are needed the "idle" calls
714 ; are disconnected (so long as there are at least "minidle" calls still
715 ; running, of course) to make more channels available. The primary use of
716 ; this is to create a dynamic service, where idle channels are bundled through
717 ; multilink PPP, thus more efficiently utilizing combined voice/data services
718 ; than conventional fixed mappings/muxings.
720 ; Those settings cannot be changed on reload.
723 ;idleext=6999@dialout
727 ; Configure jitter buffers in Zapata (each one is 20ms, default is 4)
728 ; This is set globally, rather than per-channel.
732 ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
733 ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
734 ; ZAP channel. Defaults to "no". An enabled jitterbuffer will
735 ; be used only if the sending side can create and the receiving
736 ; side can not accept jitter. The ZAP channel can't accept jitter,
737 ; thus an enabled jitterbuffer on the receive ZAP side will always
738 ; be used if the sending side can create jitter.
740 ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
742 ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
743 ; resynchronized. Useful to improve the quality of the voice, with
744 ; big jumps in/broken timestamps, usually sent from exotic devices
745 ; and programs. Defaults to 1000.
747 ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a ZAP
748 ; channel. Two implementations are currently available - "fixed"
749 ; (with size always equals to jbmax-size) and "adaptive" (with
750 ; variable size, actually the new jb of IAX2). Defaults to fixed.
752 ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
753 ;-----------------------------------------------------------------------------------
755 ; You can define your own custom ring cadences here. You can define up to 8
756 ; pairs. If the silence is negative, it indicates where the caller ID spill is
757 ; to be placed. Also, if you define any custom cadences, the default cadences
758 ; will be turned off.
760 ; This setting is global, rather than per-channel. It will not update on
763 ; Syntax is: cadence=ring,silence[,ring,silence[...]]
765 ; These are the default cadences:
767 ;cadence=125,125,2000,-4000
768 ;cadence=250,250,500,1000,250,250,500,-4000
769 ;cadence=125,125,125,125,125,-4000
770 ;cadence=1000,500,2500,-5000
772 ; Each channel consists of the channel number or range. It inherits the
773 ; parameters that were specified above its declaration.
775 ; For GR-303, CRV's are created like channels except they must start with the
776 ; trunk group followed by a colon, e.g.:
782 ;callerid="Green Phone"<(256) 428-6121>
784 ;callerid="Black Phone"<(256) 428-6122>
786 ;callerid="CallerID Phone" <(630) 372-1564>
788 ;callerid="Pac Tel Phone" <(256) 428-6124>
790 ;callerid="Uniden Dead" <(256) 428-6125>
792 ;callerid="Cortelco 2500" <(256) 428-6126>
794 ;callerid="Main TA 750" <(256) 428-6127>
797 ; For example, maybe we have some other channels which start out in a
798 ; different context and use E & M signalling instead.
807 ; All those in group 0 I'll use for outgoing calls
809 ; Strip most significant digit (9) before sending
819 ;callerid="Joe Schmoe" <(256) 428-6131>
821 ;callerid="Megan May" <(256) 428-6132>
823 ;callerid="Suzy Queue" <(256) 428-6233>
825 ;callerid="Larry Moe" <(256) 428-6234>
828 ; Sample PRI (CPE) config: Specify the switchtype, the signalling as either
829 ; pri_cpe or pri_net for CPE or Network termination, and generally you will
830 ; want to create a single "group" for all channels of the PRI.
832 ; switchtype cannot be changed on a reload.
834 ; switchtype = national
835 ; signalling = pri_cpe
841 ; Used for distinctive ring support for x100p.
842 ; You can see the dringX patterns is to set any one of the dringXcontext fields
843 ; and they will be printed on the console when an inbound call comes in.
845 ; dringXrange is used to change the acceptable ranges for "tone offsets". Defaults to 10.
846 ; Note: a range of 0 is NOT what you might expect - it instead forces it to the default.
847 ; A range of -1 will force it to always match.
848 ; Anything lower than -1 would presumably cause it to never match.
851 ;dring1context=internal1
854 ;dring2context=internal2
856 ; If no pattern is matched here is where we go.
860 ; ---------------- Options for use with signalling=ss7 -----------------
861 ; None of them can be changed by a reload.
863 ; Variant of SS7 signalling:
864 ; Options are itu and ansi
867 ; SS7 Called Nature of Address Indicator
870 ; subscriber: Subscriber
872 ; international: International
873 ; dynamic: Dynamically selects the appropriate dialplan
875 ;ss7_called_nai=dynamic
877 ; SS7 Calling Nature of Address Indicator
880 ; subscriber: Subscriber
882 ; international: International
883 ; dynamic: Dynamically selects the appropriate dialplan
885 ;ss7_calling_nai=dynamic
888 ; sample 1 for Germany
889 ;ss7_internationalprefix = 00
890 ;ss7_nationalprefix = 0
891 ;ss7_subscriberprefix =
895 ; All settings apply to linkset 1
898 ; Point code of the linkset. For ITU, this is the decimal number
899 ; format of the point code. For ANSI, this can either be in decimal
900 ; number format or in the xxx-xxx-xxx format
903 ; Point code of node adjacent to this signalling link (Possibly the STP between you and
904 ; your destination). Point code format follows the same rules as above.
907 ; Default point code that you would like to assign to outgoing messages (in case of
908 ; routing through STPs, or using A links). Point code format follows the same rules
912 ; Begin CIC (Circuit indication codes) count with this number
915 ; What the MTP3 network indicator bits should be set to. Choices are
916 ; national, national_spare, international, international_spare
917 ;networkindicator=international
919 ; First signalling channel
922 ; Channels to associate with CICs on this linkset
925 ; For more information on setting up SS7, see the README file in libss7 or
926 ; the doc/ss7.txt file in the Asterisk source tree.
927 ; ----------------- SS7 Options ----------------------------------------
929 ; Configuration Sections
930 ; ~~~~~~~~~~~~~~~~~~~~~~
931 ; You can also configure channels in a separate zapata.conf section. In
932 ; this case the keyword 'channel' is not used. Instead the keyword
933 ; 'zapchan' is used (as in users.conf) - configuration is only processed
934 ; in a section where the keyword zapchan is used. It will only be
935 ; processed in the end of the section. Thus the following section:
942 ; Is somewhat equivalent to the following snippet in the section
949 ; When starting a new section almost all of the configuration values are
950 ; copied from their values at the end of the section [channels] in
951 ; zapata.conf and [general] in users.conf - one section's configuration
952 ; does not affect another one's.
954 ; Instead of letting common configuration values "slide through" you can
955 ; use configuration templates to easily keep the common part in one
956 ; place and override where needed.
963 ;threewaycalling = yes
966 ;faxdetect = incoming
970 ;callerid = My Name <501>
971 ;mailbox = 501@mailboxes
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