rename zapata.conf.sample to chan_dahdi.conf.sample

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@126675 f38db490-d61c-443f-a65b-d21fe96a405b

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diff --git a/configs/zapata.conf.sample b/configs/zapata.conf.sample
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-;
-; DAHDI telephony
-;
-; Configuration file
-;
-; You need to restart Asterisk to re-configure the DAHDI channel
-; CLI> reload chan_dahdi.so 
-;		will reload the configuration file,
-;		but not all configuration options are 
-; 		re-configured during a reload (signalling, as well as
-;               PRI and SS7-related settings cannot be changed on a
-;               reload.
-; 
-; This file documents many configuration variables.  Normally unless you
-; know what a variable means or that it should be changed, there's no
-; reason to unrem lines.
-;
-; remmed-out examples below (those lines that begin with a ';' but no
-; space afterwards) typically show a value that is not the defauult value,
-; but would make sense under cetain circumstances. The default values
-; are usually sane. Thus you should typically not touch them unless you 
-; know what they mean or you know you should change them.
-
-
-[trunkgroups]
-;
-; Trunk groups are used for NFAS or GR-303 connections.
-;
-; Group: Defines a trunk group.  
-;        trunkgroup => <trunkgroup>,<dchannel>[,<backup1>...]
-;
-;        trunkgroup  is the numerical trunk group to create
-;        dchannel    is the DAHDI channel which will have the 
-;                    d-channel for the trunk.
-;        backup1     is an optional list of backup d-channels.
-;
-;trunkgroup => 1,24,48
-;trunkgroup => 1,24
-;
-; Spanmap: Associates a span with a trunk group
-;        spanmap => <dahdispan>,<trunkgroup>[,<logicalspan>]
-;
-;        dahdispan     is the DAHDI span number to associate
-;        trunkgroup  is the trunkgroup (specified above) for the mapping
-;        logicalspan is the logical span number within the trunk group to use.
-;                    if unspecified, no logical span number is used.
-;
-;spanmap => 1,1,1
-;spanmap => 2,1,2
-;spanmap => 3,1,3
-;spanmap => 4,1,4
-
-[channels]
-;
-; Default language
-;
-;language=en
-;
-; Context for calls. Defaults to 'default'
-;
-;context=incoming
-;
-; Switchtype:  Only used for PRI.
-;
-; national:	  National ISDN 2 (default)
-; dms100:	  Nortel DMS100
-; 4ess:           AT&T 4ESS
-; 5ess:	          Lucent 5ESS
-; euroisdn:       EuroISDN (common in Europe)
-; ni1:            Old National ISDN 1
-; qsig:           Q.SIG
-;
-;switchtype=euroisdn
-;
-; Some switches (AT&T especially) require network specific facility IE
-; supported values are currently 'none', 'sdn', 'megacom', 'tollfreemegacom', 'accunet'
-;
-; nsf cannot be changed on a reload.
-;
-;nsf=none
-;
-; PRI Dialplan: The ISDN-level Type Of Number (TON) or numbering plan, used for
-; the dialed number.  For most installations, leaving this as 'unknown' (the
-; default) works in the most cases.  In some very unusual circumstances, you
-; may need to set this to 'dynamic' or 'redundant'.  Note that if you set one
-; of the others, you will be unable to dial another class of numbers.  For
-; example, if you set 'national', you will be unable to dial local or
-; international numbers.
-;
-; PRI Local Dialplan:  Only RARELY used for PRI (sets the calling number's 
-; numbering plan).  In North America, the typical use is sending the 10 digit
-; callerID number and setting the prilocaldialplan to 'national' (the default).
-; Only VERY rarely will you need to change this.
-;
-; Neither pridialplan nor prilocaldialplan can be changed on reload.
-;
-; unknown:        Unknown
-; private:        Private ISDN
-; local:          Local ISDN
-; national:       National ISDN
-; international:  International ISDN
-; dynamic:        Dynamically selects the appropriate dialplan
-; redundant:      Same as dynamic, except that the underlying number is not 
-;                 changed (not common)
-;
-;pridialplan=unknown
-;prilocaldialplan=national
-; 
-; pridialplan may be also set at dialtime, by prefixing the dialled number with
-; one of the following letters:
-; U - Unknown
-; I - International
-; N - National
-; L - Local (Net Specific)
-; S - Subscriber
-; V - Abbreviated
-; R - Reserved (should probably never be used but is included for completeness)
-;
-; Additionally, you may also set the following NPI bits (also by prefixing the
-; dialled string with one of the following letters):
-; u - Unknown
-; e - E.163/E.164 (ISDN/telephony)
-; x - X.121 (Data)
-; f - F.69 (Telex)
-; n - National
-; p - Private
-; r - Reserved (should probably never be used but is included for completeness)
-;
-; You may also set the prilocaldialplan in the same way, but by prefixing the
-; Caller*ID Number, rather than the dialled number.  Please note that telcos
-; which require this kind of additional manipulation of the TON/NPI are *rare*.
-; Most telco PRIs will work fine simply by setting pridialplan to unknown or
-; dynamic.
-;
-;
-; PRI caller ID prefixes based on the given TON/NPI (dialplan)
-; This is especially needed for EuroISDN E1-PRIs
-; 
-; None of the prefix settings can be changed on reload.
-;
-; sample 1 for Germany 
-;internationalprefix = 00
-;nationalprefix = 0
-;localprefix = 0711
-;privateprefix = 07115678
-;unknownprefix = 
-;
-; sample 2 for Germany 
-;internationalprefix = +
-;nationalprefix = +49
-;localprefix = +49711
-;privateprefix = +497115678
-;unknownprefix = 
-;
-; PRI resetinterval: sets the time in seconds between restart of unused
-; B channels; defaults to 'never'.
-;
-;resetinterval = 3600 
-;
-; Overlap dialing mode (sending overlap digits)
-; Cannot be changed on a reload.
-;
-;overlapdial=yes
-;
-; PRI Out of band indications.
-; Enable this to report Busy and Congestion on a PRI using out-of-band
-; notification. Inband indication, as used by Asterisk doesn't seem to work
-; with all telcos.
-; 
-; outofband:      Signal Busy/Congestion out of band with RELEASE/DISCONNECT
-; inband:         Signal Busy/Congestion using in-band tones (default)
-;
-; priindication cannot be changed on a reload.
-;
-;priindication = outofband
-;
-; If you need to override the existing channels selection routine and force all
-; PRI channels to be marked as exclusively selected, set this to yes.
-;
-; priexclusive cannot be changed on a reload.
-;
-;priexclusive = yes
-;
-; ISDN Timers
-; All of the ISDN timers and counters that are used are configurable.  Specify
-; the timer name, and its value (in ms for timers).
-; K:    Layer 2 max number of outstanding unacknowledged I frames (default 7)
-; N200: Layer 2 max number of retransmissions of a frame (default 3)
-; T200: Layer 2 max time before retransmission of a frame (default 1000 ms)
-; T203: Layer 2 max time without frames being exchanged (default 10000 ms)
-; T305: Wait for DISCONNECT acknowledge (default 30000 ms)
-; T308: Wait for RELEASE acknowledge (default 4000 ms)
-; T309: Maintain active calls on Layer 2 disconnection (default -1, 
-;       Asterisk clears calls)
-;       EuroISDN: 6000 to 12000 ms, according to (N200 + 1) x T200 + 2s
-;       May vary in other ISDN standards (Q.931 1993 : 90000 ms)
-; T313: Wait for CONNECT acknowledge, CPE side only (default 3000 ms)
-;
-;pritimer => t200,1000
-;pritimer => t313,4000
-;
-; To enable transmission of facility-based ISDN supplementary services (such
-; as caller name from CPE over facility), enable this option.
-; Cannot be changed on a reload.
-;
-;facilityenable = yes
-;
-; pritimer cannot be changed on a reload.
-;
-; Signalling method. The default is "auto". Valid values:
-; auto:           Use the current value from DAHDI.
-; em:             E & M
-; em_e1:          E & M E1
-; em_w:           E & M Wink
-; featd:          Feature Group D (The fake, Adtran style, DTMF)
-; featdmf:        Feature Group D (The real thing, MF (domestic, US))
-; featdmf_ta:     Feature Group D (The real thing, MF (domestic, US)) through
-;                 a Tandem Access point
-; featb:          Feature Group B (MF (domestic, US))
-; fgccama	  Feature Group C-CAMA (DP DNIS, MF ANI)
-; fgccamamf	  Feature Group C-CAMA MF (MF DNIS, MF ANI)
-; 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
-; gr303fxoks_net: GR-303 Signalling, FXO Loopstart, Network side
-; gr303fxsks_cpe: GR-303 Signalling, FXS Loopstart, CPE 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))
-; e911:           E911 (MF) style signalling
-; ss7:            Signalling System 7
-;
-; 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)
-; ss7:            Signalling System 7
-;
-; signalling of a channel can not be changed on a reload.
-;
-;signalling=fxo_ls
-;
-; If you have an outbound signalling format that is different from format
-; specified above (but compatible), you can specify outbound signalling format,
-; (see below). The 'signalling' format specified will be the inbound signalling
-; format. If you only specify 'signalling', then it will be the format for
-; both inbound and outbound.
-; 
-; outsignalling can only be one of: 
-;   em, em_e1, em_w, sf, sf_w, sf_featd, sf_featdmf, sf_featb, featd,
-;   featdmf, featdmf_ta, e911, fgccama, fgccamamf
-; 
-; outsignalling cannot be changed on a reload.
-;
-;signalling=featdmf
-;
-;outsignalling=featb
-;
-; For Feature Group D Tandem access, to set the default CIC and OZZ use these
-; parameters (Will not be updated on reload):
-;
-;defaultozz=0000
-;defaultcic=303
-;
-; A variety of timing parameters can be specified as well
-; The default values for those are "-1", which is to use the
-; compile-time defaults of the DAHDI kernel modules. The timing
-; parameters, (with the standard default from DAHDI):
-;
-;    prewink:     Pre-wink time (default 50ms)
-;    preflash:    Pre-flash time (default 50ms)
-;    wink:        Wink time (default 150ms)
-;    flash:       Flash time (default 750ms)
-;    start:       Start time (default 1500ms)
-;    rxwink:      Receiver wink time (default 300ms)
-;    rxflash:     Receiver flashtime (default 1250ms)
-;    debounce:    Debounce timing (default 600ms)
-;
-; None of them will update on a reload.
-;
-; How long generated tones (DTMF and MF) will be played on the channel
-; (in milliseconds). 
-;
-; This is a global, rather than a per-channel setting. It will not be 
-; updated on a reload.
-;
-;toneduration=100
-;
-; Whether or not to do distinctive ring detection on FXO lines:
-;
-;usedistinctiveringdetection=yes
-;
-; enable dring detection after caller ID for those countries like Australia
-; where the ring cadence is changed *after* the caller ID spill:
-;
-;distinctiveringaftercid=yes	
-;
-; Whether or not to use caller ID:
-;
-usecallerid=yes
-;
-; Hide the name part and leave just the number part of the caller ID
-; string. Only applies to PRI channels.
-;hidecalleridname=yes
-;
-; Type of caller ID signalling in use
-;     bell     = bell202 as used in US (default)
-;     v23      = v23 as used in the UK
-;     v23_jp   = v23 as used in Japan
-;     dtmf     = DTMF as used in Denmark, Sweden and Netherlands
-;     smdi     = Use SMDI for caller ID.  Requires SMDI to be enabled (usesmdi).
-;
-;cidsignalling=v23
-;
-; What signals the start of caller ID
-;     ring        = a ring signals the start (default)
-;     polarity    = polarity reversal signals the start
-;     polarity_IN = polarity reversal signals the start, for India, 
-;                   for dtmf dialtone detection; using DTMF.
-;                   (see doc/India-CID.txt)
-;
-;cidstart=polarity
-;
-; Whether or not to hide outgoing caller ID (Override with *67 or *82)
-; (If your dialplan doesn't catch it)
-;
-;hidecallerid=yes
-;
-; The following option enables receiving MWI on FXO lines.  The default
-; value is no.  When this is enabled, and MWI notification indicates on or off,
-; the script specified by the mwimonitornotify option is executed.  Also, an
-; internal Asterisk MWI event will be generated so that any other part of
-; Asterisk that cares about MWI state changes will get notified, just as if
-; the state change came from app_voicemail. The energy level that must be seen
-; before starting the MWI detection process can be set with 'mwilevel'.
-;
-;mwimonitor=no
-;mwilevel=512
-;
-; This option is used in conjunction with mwimonitor.  This will get executed
-; when incoming MWI state changes.  The script is passed 2 arguments.  The
-; first is the corresponding mailbox, and the second is 1 or 0, indicating if
-; there are messages waiting or not.
-;
-;mwimonitornotify=/usr/local/bin/dahdinotify.sh
-;
-; Whether or not to enable call waiting on internal extensions
-; With this set to 'yes', busy extensions will hear the call-waiting
-; tone, and can use hook-flash to switch between callers. The Dial()
-; app will not return the "BUSY" result for extensions.
-;
-callwaiting=yes
-;
-; Whether or not restrict outgoing caller ID (will be sent as ANI only, not
-; available for the user)
-; Mostly use with FXS ports
-;
-;restrictcid=no
-;
-; Whether or not use the caller ID presentation for the outgoing call that the
-; calling switch is sending.
-; See README.callingpres. FIXME: file no longer exists.
-;
-usecallingpres=yes
-;
-; Some countries (UK) have ring tones with different ring tones (ring-ring),
-; which means the caller ID needs to be set later on, and not just after
-; the first ring, as per the default (1). 
-;
-;sendcalleridafter = 2
-;
-;
-; Support caller ID on Call Waiting
-;
-callwaitingcallerid=yes
-;
-; Support three-way calling
-;
-threewaycalling=yes
-;
-; For FXS ports (either direct analog or over T1/E1):
-;   Support flash-hook call transfer (requires three way calling)
-;   Also enables call parking (overrides the 'canpark' parameter)
-;
-; For digital ports using ISDN PRI protocols:
-;   Support switch-side transfer (called 2BCT, RLT or other names)
-;   This setting must be enabled on both ports involved, and the
-;   'facilityenable' setting must also be enabled to allow sending
-;   the transfer to the ISDN switch, since it sent in a FACILITY
-;   message.
-;
-transfer=yes
-;
-; Allow call parking
-; ('canpark=no' is overridden by 'transfer=yes')
-;
-canpark=yes
-;
-; Support call forward variable
-;
-cancallforward=yes
-;
-; Whether or not to support Call Return (*69, if your dialplan doesn't
-; catch this first)
-;
-callreturn=yes
-;
-; Stutter dialtone support: If a mailbox is specified without a voicemail 
-; context, then when voicemail is received in a mailbox in the default 
-; voicemail context in voicemail.conf, taking the phone off hook will cause a
-; stutter dialtone instead of a normal one. 
-;
-; If a mailbox is specified *with* a voicemail context, the same will result
-; if voicemail received in mailbox in the specified voicemail context.
-;
-; for default voicemail context, the example below is fine:
-;
-;mailbox=1234
-;
-; for any other voicemail context, the following will produce the stutter tone:
-;
-;mailbox=1234@context 
-;
-; 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.
-;
-; Note that when setting the number of taps, the number 256 does not translate
-; to 256 ms of echo cancellation.  echocancel=256 means 256 / 8 = 32 ms.
-;
-; Note that if any of your DAHDI cards have hardware echo cancellers,
-; then this setting only turns them on and off; numeric settings will
-; be treated as "yes". There are no special settings required for
-; hardware echo cancellers; when present and enabled in their kernel
-; modules, they take precedence over the software echo canceller compiled
-; into DAHDI automatically.
-;
-;
-echocancel=yes
-;
-; As of Zaptel 1.4.8, some DAHDI echo cancellers (software and hardware)
-; support adjustable parameters; these parameters can be supplied as
-; additional options to the 'echocancel' setting. Note that Asterisk
-; does not attempt to validate the parameters or their values, so if you
-; supply an invalid parameter you will not know the specific reason it
-; failed without checking the kernel message log for the error(s)
-; put there by DAHDI.
-;
-;echocancel=128,param1=32,param2=0,param3=14
-;
-; Generally, it is not necessary (and in fact undesirable) to echo cancel when
-; the circuit path is entirely TDM.  You may, however, change this behavior
-; by enabling the echo canceller during pure TDM bridging below.
-;
-echocancelwhenbridged=yes
-;
-; 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
-; DAHDI 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.  Value may be "yes", "no", or a number of
-; milliseconds to delay before training (default = 400)
-;
-; WARNING:  In some cases this option can make echo worse!  If you are
-; trying to debug an echo problem, it is worth checking to see if your echo
-; is better with the option set to yes or no.  Use whatever setting gives
-; the best results.
-;
-; Note that these parameters do not apply to hardware echo cancellers.
-;
-;echotraining=yes
-;echotraining=800
-;
-; 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.
-;
-;relaxdtmf=yes
-;
-; You may also set the default receive and transmit gains (in dB)
-;
-; Gain Settings: increasing / decreasing the volume level on a channel.
-;                The values are in db (decibells). A positive number
-;                increases the volume level on a channel, and a
-;                negavive value decreases volume level.
-;
-;                There are several independent gain settings:
-;   rxgain: gain for the rx (receive - into Asterisk) channel. Default: 0.0
-;   txgain: gain for the tx (transmit - out of Asterisk Asterisk) channel. 
-;           Default: 0.0
-;   cid_rxgain: set the gain just for the caller ID sounds Asterisk
-;               emits. Default: 5.0 .
-
-;rxgain=2.0
-;txgain=3.0
-;
-; Logical groups can be assigned to allow outgoing roll-over.  Groups range
-; from 0 to 63, and multiple groups can be specified. By default the
-; channel is not a member of any group.
-;
-; Note that an explicit empty value for 'group' is invalid, and will not
-; override a previous non-empty one. The same applies to callgroup and
-; pickupgroup as well.
-;
-group=1
-;
-; 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.  Groups range from 0 to 63.
-;
-callgroup=1
-pickupgroup=1
-
-; Channel variable to be set for all calls from this channel
-;setvar=CHANNEL=42             
-
-;
-; Specify whether the channel should be answered immediately or if the simple
-; switch should provide dialtone, read digits, etc.
-; Note: If immediate=yes the dialplan execution will always start at extension
-; 's' priority 1 regardless of the dialed number!
-;
-;immediate=yes
-;
-; Specify whether flash-hook transfers to 'busy' channels should complete or
-; return to the caller performing the transfer (default is yes).
-;
-;transfertobusy=no
-;
-; caller ID can be set to "asreceived" or a specific number if you want to
-; override it.  Note that "asreceived" only applies to trunk interfaces.
-; fullname sets just the 
-;
-; fullname: sets just the name part.
-; cid_number: sets just the number part: 
-;
-;callerid = 123456
-;
-;callerid = My Name <2564286000>
-; Which can also be written as:
-;cid_number = 2564286000
-;fullname = My Name
-;
-;callerid = asreceived
-;
-; should we use the caller ID from incoming call on DAHDI transfer?
-;
-;useincomingcalleridondahditransfer = yes
-;
-; AMA flags affects the recording of Call Detail Records.  If specified
-; it may be 'default', 'omit', 'billing', or 'documentation'.
-;
-;amaflags=default
-;
-; Channels may be associated with an account code to ease
-; billing
-;
-;accountcode=lss0101
-;
-; ADSI (Analog Display Services Interface) can be enabled on a per-channel
-; basis if you have (or may have) ADSI compatible CPE equipment
-;
-;adsi=yes
-;
-; SMDI (Simplified Message Desk Interface) can be enabled on a per-channel
-; basis if you would like that channel to behave like an SMDI message desk.
-; The SMDI port specified should have already been defined in smdi.conf.  The
-; default port is /dev/ttyS0.
-;
-;usesmdi=yes
-;smdiport=/dev/ttyS0
-;
-; 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.  This enables listening for
-; the beep-beep busy pattern.
-;
-;busydetect=yes
-;
-; If busydetect is enabled, it is also possible to specify how many busy tones
-; to wait for before hanging up.  The default is 3, but it might be
-; safer to set to 6 or even 8.  Mind that the higher the number, the more
-; time that will be needed to hangup a channel, but lowers the probability
-; that you will get random hangups.
-;
-;busycount=6
-;
-; If busydetect is enabled, it is also possible to specify the cadence of your
-; busy signal.  In many countries, it is 500msec on, 500msec off.  Without
-; busypattern specified, we'll accept any regular sound-silence pattern that
-; repeats <busycount> times as a busy signal.  If you specify busypattern,
-; then we'll further check the length of the sound (tone) and silence, which
-; will further reduce the chance of a false positive.
-;
-;busypattern=500,500
-;
-; NOTE: In make menuselect, you'll find further options to tweak the busy
-; detector.  If your country has a busy tone with the same length tone and
-; silence (as many countries do), consider enabling the
-; BUSYDETECT_COMPARE_TONE_AND_SILENCE option.
-;
-; To further detect which hangup tone your telco provider is sending, it is
-; useful to use the ztmonitor utility to record the audio that main/dsp.c
-; is receiving after the caller hangs up.
-;
-; Use a polarity reversal to mark when a outgoing call is answered by the
-; remote party.
-;
-;answeronpolarityswitch=yes
-;
-; In some countries, a polarity reversal is used to signal the disconnect of a
-; phone line.  If the hanguponpolarityswitch option is selected, the call will
-; be considered "hung up" on a polarity reversal.
-;
-;hanguponpolarityswitch=yes
-;
-; polarityonanswerdelay: minimal time period (ms) between the answer 
-;                        polarity switch and hangup polarity switch. 
-;                        (default: 600ms)
-;
-; 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.
-;
-; Few zones are supported at the time of this writing, but may be selected
-; with "progzone".
-;
-; progzone also affects the pattern used for buzydetect (unless
-; busypattern is set explicitly). The possible values are: 
-;   us (default)
-;   ca (alias for 'us')
-;   cr (Costa Rica)
-;   br (Brazil, alias for 'cr')
-;   uk
-;
-; This feature can also easily detect false hangups. The symptoms of this is
-; being disconnected in the middle of a call for no reason.
-;
-;callprogress=yes
-;progzone=uk
-;
-; Set the tonezone. Equivalent of the defaultzone settings in
-; /etc/dahdi.conf . This sets the tone zone by number.
-; Note that you'd still need to load tonezones (loadzone in dahdi.conf).
-; The default is -1: not to set anything.
-;tonezone = 0 ; 0 is US
-;
-; FXO (FXS signalled) devices must have a timeout to determine if there was a
-; hangup before the line was answered.  This value can be tweaked to shorten
-; how long it takes before DAHDI considers a non-ringing line to have hungup.
-;
-; ringtimeout will not update on a reload.
-;
-;ringtimeout=8000
-;
-; For FXO (FXS signalled) devices, whether to use pulse dial instead of DTMF
-; Pulse digits from phones (FXS devices, FXO signalling) are always
-; detected.
-;
-;pulsedial=yes
-;
-; For fax detection, uncomment one of the following lines.  The default is *OFF*
-;
-;faxdetect=both
-;faxdetect=incoming
-;faxdetect=outgoing
-;faxdetect=no
-;
-; This option specifies a preference for which music on hold class this channel
-; should listen to when put on hold if the music class has not been set on the
-; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
-; channel putting this one on hold did not suggest a music class.
-;
-; If this option is set to "passthrough", then the hold message will always be
-; passed through as signalling instead of generating hold music locally. This
-; setting is only valid when used on a channel that uses digital signalling.
-;
-;mohinterpret=default
-;
-; This option specifies which music on hold class to suggest to the peer channel
-; when this channel places the peer on hold. 
-;
-;mohsuggest=default
-;
-; PRI channels can have an idle extension and a minunused number.  So long as
-; at least "minunused" channels are idle, chan_dahdi 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.
-;
-; Those settings cannot be changed on reload.
-;
-;idledial=6999
-;idleext=6999@dialout
-;minunused=2
-;minidle=1
-;
-; Configure jitter buffers in DAHDI (each one is 20ms, default is 4)
-; This is set globally, rather than per-channel.
-;
-;jitterbuffers=4
-;
-;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
-; jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of a
-                              ; DAHDI channel. Defaults to "no". An enabled jitterbuffer will
-                              ; be used only if the sending side can create and the receiving
-                              ; side can not accept jitter. The DAHDI channel can't accept jitter,
-                              ; thus an enabled jitterbuffer on the receive DAHDI side will always
-                              ; be used if the sending side can create jitter.
-
-; jbmaxsize = 200             ; Max length of the jitterbuffer in milliseconds.
-
-; jbresyncthreshold = 1000    ; Jump in the frame timestamps over which the jitterbuffer is
-                              ; resynchronized. Useful to improve the quality of the voice, with
-                              ; big jumps in/broken timestamps, usually sent from exotic devices
-                              ; and programs. Defaults to 1000.
-
-; jbimpl = fixed              ; Jitterbuffer implementation, used on the receiving side of a DAHDI
-                              ; channel. Two implementations are currently available - "fixed"
-                              ; (with size always equals to jbmax-size) and "adaptive" (with
-                              ; variable size, actually the new jb of IAX2). Defaults to fixed.
-
-; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
-;-----------------------------------------------------------------------------------
-;
-; You can define your own custom ring cadences here.  You can define up to 8
-; pairs.  If the silence is negative, it indicates where the caller ID spill is
-; to be placed.  Also, if you define any custom cadences, the default cadences
-; will be turned off.
-;
-; This setting is global, rather than per-channel. It will not update on
-; a reload.
-;
-; Syntax is:  cadence=ring,silence[,ring,silence[...]]
-;
-; These are the default cadences:
-;
-;cadence=125,125,2000,-4000
-;cadence=250,250,500,1000,250,250,500,-4000
-;cadence=125,125,125,125,125,-4000
-;cadence=1000,500,2500,-5000
-;
-; Each channel consists of the channel number or range.  It inherits the
-; parameters that were specified above its declaration.
-;
-; For GR-303, CRV's are created like channels except they must start with the
-; trunk group followed by a colon, e.g.: 
-;
-; crv => 1:1
-; crv => 2:1-2,5-8
-;
-;
-;callerid="Green Phone"<(256) 428-6121>
-;channel => 1
-;callerid="Black Phone"<(256) 428-6122>
-;channel => 2
-;callerid="CallerID Phone" <(630) 372-1564>
-;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.
-;
-;context=remote
-;sigalling=em
-;channel => 15
-;channel => 16
-
-;signalling=em_w
-;
-; All those in group 0 I'll use for outgoing calls
-;
-; Strip most significant digit (9) before sending
-;
-;stripmsd=1
-;callerid=asreceived
-;group=0
-;signalling=fxs_ls
-;channel => 45
-
-;signalling=fxo_ls
-;group=1
-;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 cannot be changed on a reload.
-;
-; switchtype = national
-; signalling = pri_cpe
-; group = 2
-; channel => 1-23
-
-;
-
-;  Used for distinctive 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.
-;
-;  dringXrange is used to change the acceptable ranges for "tone offsets".  Defaults to 10.
-;  Note: a range of 0 is NOT what you might expect - it instead forces it to the default.
-;  A range of -1 will force it to always match.
-;  Anything lower than -1 would presumably cause it to never match.
-;
-;dring1=95,0,0 
-;dring1context=internal1 
-;dring1range=10
-;dring2=325,95,0 
-;dring2context=internal2 
-;dring2range=10
-; If no pattern is matched here is where we go.
-;context=default
-;channel => 1 
-
-; ---------------- Options for use with signalling=ss7 -----------------
-; None of them can be changed by a reload.
-;
-; Variant of SS7 signalling:
-; Options are itu and ansi
-;ss7type = itu
-
-; SS7 Called Nature of Address Indicator
-;
-; unknown:        Unknown
-; subscriber:     Subscriber
-; national:	  National
-; international:  International
-; dynamic:	  Dynamically selects the appropriate dialplan
-;
-;ss7_called_nai=dynamic
-;
-; SS7 Calling Nature of Address Indicator
-;
-; unknown:        Unknown
-; subscriber:     Subscriber
-; national:	  National
-; international:  International
-; dynamic:	  Dynamically selects the appropriate dialplan
-;
-;ss7_calling_nai=dynamic
-;
-; 
-; sample 1 for Germany 
-;ss7_internationalprefix = 00
-;ss7_nationalprefix = 0
-;ss7_subscriberprefix = 
-;ss7_unknownprefix = 
-;
-
-; All settings apply to linkset 1
-;linkset = 1
-
-; Point code of the linkset.  For ITU, this is the decimal number
-; format of the point code.  For ANSI, this can either be in decimal
-; number format or in the xxx-xxx-xxx format
-;pointcode = 1
-
-; Point code of node adjacent to this signalling link (Possibly the STP between you and
-; your destination).  Point code format follows the same rules as above.
-;adjpointcode = 2
-
-; Default point code that you would like to assign to outgoing messages (in case of
-; routing through STPs, or using A links).  Point code format follows the same rules
-; as above.
-;defaultdpc = 3
-
-; Begin CIC (Circuit indication codes) count with this number
-;cicbeginswith = 1
-
-; What the MTP3 network indicator bits should be set to.  Choices are
-; national, national_spare, international, international_spare
-;networkindicator=international
-
-; First signalling channel
-;sigchan = 48
-
-; Channels to associate with CICs on this linkset
-;channel = 25-47
-;
-; For more information on setting up SS7, see the README file in libss7 or
-; the doc/ss7.txt file in the Asterisk source tree.
-; ----------------- SS7 Options ----------------------------------------
-
-; Configuration Sections
-; ~~~~~~~~~~~~~~~~~~~~~~
-; You can also configure channels in a separate dahdi.conf section. In
-; this case the keyword 'channel' is not used. Instead the keyword
-; 'dahdichan' is used (as in users.conf) - configuration is only processed
-; in a section where the keyword dahdichan is used. It will only be
-; processed in the end of the section. Thus the following section:
-;
-;[phones]
-;echocancel = 64
-;dahdichan = 1-8
-;group = 1
-;
-; Is somewhat equivalent to the following snippet in the section
-; [channels]:
-;
-;echocancel = 64
-;group = 1
-;channel => 1-8
-;
-; When starting a new section almost all of the configuration values are
-; copied from their values at the end of the section [channels] in
-; dahdi.conf and [general] in users.conf - one section's configuration
-; does not affect another one's.
-;
-; Instead of letting common configuration values "slide through" you can 
-; use configuration templates to easily keep the common part in one
-; place and override where needed.
-;
-;[phones](!)
-;echocancel = yes
-;group = 0,4
-;callgroup = 3
-;pickupgroup = 3
-;threewaycalling = yes
-;transfer = yes
-;context = phones
-;faxdetect = incoming
-;
-;[phone-1](phones)
-;dahdichan = 1
-;callerid = My Name <501>
-;mailbox = 501@mailboxes
-;
-;
-;[fax](phones)
-;dahdichan = 2
-;faxdetect = no
-;context = fax
-;
-;[phone-3](phones)
-;dahdichan = 3
-;pickupgroup = 3,4