Asterisk config zapata.conf.sample

; Zapata telephony interface
; Configuration file

; Default language
; Default context
; Switchtype:  Only used for PRI.
; national:       National ISDN 2 (default)
; dms100:         Nortel DMS100
; 4ess:           AT&T 4ESS
; 5ess:           Lucent 5ESS
; euroisdn:       EuroISDN
; ni1:            Old National ISDN 1
; PRI Dialplan:  Only RARELY used for PRI.
; unknown:        Unknown
; private:        Private ISDN
; local:          Local ISDN
; national:       National ISDN
; international:  International ISDN
; note: to connect to at&t LD PRI
;                 switchtype => 4ess
;                 nsf=megacom
; Overlap dialing mode (sending overlap digits)
; Signalling method (default is fxs).  Valid values:
; em:      E & M
; em_w:    E & M Wink
; featd:   Feature Group D (The fake, Adtran style, DTMF)
; featdmf: Feature Group D (The real thing, MF (domestic, US))
; featb:   Feature Group B (MF (domestic, US))
; fxs_ls:  FXS (Loop Start)
; fxs_gs:  FXS (Ground Start)
; fxs_ks:  FXS (Kewl Start)
; fxo_ls:  FXO (Loop Start)
; fxo_gs:  FXO (Ground Start)
; fxo_ks:  FXO (Kewl Start)
; pri_cpe: PRI signalling, CPE side
; pri_net: PRI signalling, Network side
; sf:         SF (Inband Tone) Signalling
; sf_w:       SF Wink
; sf_featd:   SF Feature Group D (The fake, Adtran style, DTMF)
; sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US))
; sf_featb:   SF Feature Group B (MF (domestic, US))
; The following are used for Radio interfaces:
; fxs_rx:  Receive audio/COR on an FXS kewlstart interface (FXO at the channel bank)
; fxs_tx:  Transmit audio/PTT on an FXS loopstart interface (FXO at the channel bank)
; fxo_rx:  Receive audio/COR on an FXO loopstart interface (FXS at the channel bank)
; fxo_tx:  Transmit audio/PTT on an FXO groundstart interface (FXS at the channel bank)
; em_rx:   Receive audio/COR on an E&M interface (1-way)
; em_tx:   Transmit audio/PTT on an E&M interface (1-way)
; em_txrx: Receive audio/COR AND Transmit audio/PTT on an E&M interface (2-way)
; em_rxtx: same as em_txrx (for our dyslexic friends)
; sf_rx:   Receive audio/COR on an SF interface (1-way)
; sf_tx:   Transmit audio/PTT on an SF interface (1-way)
; sf_txrx: Receive audio/COR AND Transmit audio/PTT on an SF interface (2-way)
; sf_rxtx: same as sf_txrx (for our dyslexic friends)
; Whether or not to do distinctive ring detection on FXO lines

; Whether or not to use caller ID
; Whether or not to hide outgoing caller ID (Override with *67 or *82)
; Whether or not to enable call waiting on FXO lines
; Whether or not restrict outgoing caller ID (will be sent as ANI only, not available for the user)
; Mostly use with FXS ports
; Whether or not use the caller ID presentation for the outgoing call that the calling switch is sending
; Support Caller*ID on Call Waiting
; Support three-way calling
; Support flash-hook call transfer (requires three way calling)
; Support call forward variable
; Whether or not to support Call Return (*69)
; Stutter dialtone support: If a mailbox is specified, then when voicemail
; is received in that mailbox, taking the phone off hook will cause
; a stutter dialtone instead of a normal one
; Enable echo cancellation 
; Use either "yes", "no", or a power of two from 32 to 256 if you wish
; to actually set the number of taps of cancellation.
; Generally, it is not necessary (and in fact undesirable) to echo cancel
; when the circuit path is entirely TDM.  You may, however, reverse this
; behavior by enabling the echo cancel during pure TDM bridging below.
; In some cases, the echo canceller doesn't train quickly enough and there
; is echo at the beginning of the call.  Enabling echo training will cause
; asterisk to briefly mute the channel, send an impulse, and use the impulse
; response to pre-train the echo canceller so it can start out with a much
; closer idea of the actual echo.
; If you are having trouble with DTMF detection, you can relax the
; DTMF detection parameters.  Relaxing them may make the DTMF detector
; more likely to have "talkoff" where DTMF is detected when it
; shouldn't be.
; You may also set the default receive and transmit gains (in dB)
; Logical groups can be assigned to allow outgoing rollover.  Groups
; range from 0 to 31, and multiple groups can be specified.
; Ring groups (a.k.a. call groups) and pickup groups.  If a phone is ringing
; and it is a member of a group which is one of your pickup groups, then
; you can answer it by picking up and dialing *8#.  For simple offices, just
; make these both the same

; Specify whether the channel should be answered immediately or
; if the simple switch should provide dialtone, read digits, etc.
; CallerID can be set to "asreceived" or a specific number
; if you want to override it.  Note that "asreceived" only
; applies to trunk interfaces.
; AMA flags affects the recording of Call Detail Records.  If specified
; it may be 'default', 'omit', 'billing', or 'documentation'.
; Channels may be associated with an account code to ease
; billing
; ADSI (Analog Display Services Interface) can be enabled on a per-channel
; basis if you have (or may have) ADSI compatible CPE equipment
; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D
; etc, it can be useful to perform busy detection either in an effort to 
; detect hangup or for detecting busies
; If busydetect is enabled, is also possible to specify how many
; busy tones to wait before hanging up. The default is 4, but
; better results can be achieved if set to 6 or even 8. Mind that
; higher the number, more time is needed to hangup a channel, but
; lower is probability to get random hangups
; On trunk interfaces (FXS) it can be useful to attempt to follow the progress
; of a call through RINGING, BUSY, and ANSWERING.   If turned on, call
; progress attempts to determine answer, busy, and ringing on phone lines.
; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers,
; so don't count on it being very accurate.  Also, it is ONLY configured for
; standard U.S. tones.  This feature can also easily detect false hangups.
; The symptoms of this is being disconnected in the middle of a call for no
; reason.
; Select which class of music to use for music on hold.  If not specified
; then the default will be used.
; PRI channels can have an idle extension and a minunused number.  So long
; as at least "minunused" channels are idle, chan_zap will try to call
; "idledial" on them, and then dump them into the PBX in the "idleext"
; extension (which is of the form exten@context).  When channels are needed
; the "idle" calls are disconnected (so long as there are at least "minidle"
; calls still running, of course) to make more channels available.  The
; primary use of this is to create a dynamic service, where idle channels
; are bundled through multilink PPP, thus more efficiently utilizing
; combined voice/data services than conventional fixed mappings/muxings.
; Configure jitter buffers in zapata (each one is 20ms, default is 4, max is 32).
; The zaptel jitterbuffers differ from IAX2 jitterbuffers in that they serve 
; to queue data that is being written to the zaptel interface if the write 
; needs to be put off for a moment. An excessive value simply wastes memory
; however too low a value will result in choppy audio leaving the zaptel 
; interface and reams of console messages along the lines of: 
;     chan_zap.c:4409 my_zt_write: Write returned -1 (Resource temporarily 
;         unavailable) on channel 1 - audio may have been lost 
; Each channel consists of the channel number or range.  It
; inherits the parameters that were specified above its declaration
;callerid="Green Phone"<(256) 428-6121>
;channel => 1
;callerid="Black Phone"<(256) 428-6122>
;channel => 2
;callerid="CallerID Phone" <(256) 428-6123>
;callerid="CallerID Phone" <(630) 372-1564>
;callerid="CallerID Phone" <(256) 704-4666>
;channel => 3
;callerid="Pac Tel Phone" <(256) 428-6124>
;channel => 4
;callerid="Uniden Dead" <(256) 428-6125>
;channel => 5
;callerid="Cortelco 2500" <(256) 428-6126>
;channel => 6
;callerid="Main TA 750" <(256) 428-6127>
;channel => 44
; For example, maybe we have some other channels
; which start out in a different context and use
; E & M signalling instead.
;channel => 15
;channel => 16

; All those in group 0 I'll use for outgoing calls
; Strip most significant digit (9) before sending
;channel => 45

;callerid="Joe Schmoe" <(256) 428-6131>
;channel => 25
;callerid="Megan May" <(256) 428-6132>
;channel => 26
;callerid="Suzy Queue" <(256) 428-6233>
;channel => 27
;callerid="Larry Moe" <(256) 428-6234>
;channel => 28
; Sample PRI (CPE) config:  Specify the switchtype, the signalling as
; either pri_cpe or pri_net for CPE or Network termination, and generally
; you will want to create a single "group" for all channels of the PRI.
; switchtype = national
; signalling = pri_cpe
; group = 2
; channel => 1-23

;  Used for distintive ring support for x100p.
;  You can see the dringX patterns is to set any one of the dringXcontext fields
;  and they will be printed on the console when an inbound call comes in.
; If no pattern is matched here is where we go.
;channel => 1

Created by: JazEzork, Last modification: Thu 04 of Sep, 2008 (14:08 UTC) by jsturtevant
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