asterisk/zapata.conf

changeset 202
f29abea29121
     1.1 --- /dev/null	Thu Jan 01 00:00:00 1970 +0000
     1.2 +++ b/asterisk/zapata.conf	Mon Apr 27 12:19:05 2009 +0200
     1.3 @@ -0,0 +1,663 @@
     1.4 +;
     1.5 +; Zapata telephony interface
     1.6 +;
     1.7 +; Configuration file
     1.8 +;
     1.9 +; You need to restart Asterisk to re-configure the Zap channel
    1.10 +; CLI> reload chan_zap.so 
    1.11 +;		will reload the configuration file,
    1.12 +;		but not all configuration options are 
    1.13 +; 		re-configured during a reload.
    1.14 +
    1.15 +
    1.16 +
    1.17 +;[trunkgroups]
    1.18 +;
    1.19 +; Trunk groups are used for NFAS or GR-303 connections.
    1.20 +;
    1.21 +; Group: Defines a trunk group.  
    1.22 +;        trunkgroup => <trunkgroup>,<dchannel>[,<backup1>...]
    1.23 +;
    1.24 +;        trunkgroup  is the numerical trunk group to create
    1.25 +;        dchannel    is the zap channel which will have the 
    1.26 +;                    d-channel for the trunk.
    1.27 +;        backup1     is an optional list of backup d-channels.
    1.28 +;
    1.29 +;trunkgroup => 1,24,48
    1.30 +;trunkgroup => 1,24
    1.31 +;
    1.32 +; Spanmap: Associates a span with a trunk group
    1.33 +;        spanmap => <zapspan>,<trunkgroup>[,<logicalspan>]
    1.34 +;
    1.35 +;        zapspan     is the zap span number to associate
    1.36 +;        trunkgroup  is the trunkgroup (specified above) for the mapping
    1.37 +;        logicalspan is the logical span number within the trunk group to use.
    1.38 +;                    if unspecified, no logical span number is used.
    1.39 +;
    1.40 +;spanmap => 1,1,1
    1.41 +;spanmap => 2,1,2
    1.42 +;spanmap => 3,1,3
    1.43 +;spanmap => 4,1,4
    1.44 +
    1.45 +;[channels]
    1.46 +;
    1.47 +; Default language
    1.48 +;
    1.49 +;language=en
    1.50 +;
    1.51 +; Default context
    1.52 +;
    1.53 +;context=default
    1.54 +;
    1.55 +; Switchtype:  Only used for PRI.
    1.56 +;
    1.57 +; national:	  National ISDN 2 (default)
    1.58 +; dms100:	  Nortel DMS100
    1.59 +; 4ess:           AT&T 4ESS
    1.60 +; 5ess:	          Lucent 5ESS
    1.61 +; euroisdn:       EuroISDN
    1.62 +; ni1:            Old National ISDN 1
    1.63 +; qsig:           Q.SIG
    1.64 +;
    1.65 +;switchtype=national
    1.66 +;
    1.67 +; Some switches (AT&T especially) require network specific facility IE
    1.68 +; supported values are currently 'none', 'sdn', 'megacom', 'tollfreemegacom', 'accunet'
    1.69 +;
    1.70 +;nsf=none
    1.71 +;
    1.72 +; PRI Dialplan:  Only RARELY used for PRI.
    1.73 +;
    1.74 +; unknown:        Unknown
    1.75 +; private:        Private ISDN
    1.76 +; local:          Local ISDN
    1.77 +; national:	  National ISDN
    1.78 +; international:  International ISDN
    1.79 +; dynamic:	  Dynamically selects the appropriate dialplan
    1.80 +;
    1.81 +;pridialplan=national
    1.82 +;
    1.83 +; PRI Local Dialplan:  Only RARELY used for PRI (sets the calling number's numbering plan)
    1.84 +;
    1.85 +; unknown:        Unknown
    1.86 +; private:        Private ISDN
    1.87 +; local:          Local ISDN
    1.88 +; national:	  National ISDN
    1.89 +; international:  International ISDN
    1.90 +; dynamic:	  Dynamically selects the appropriate dialplan
    1.91 +;
    1.92 +;prilocaldialplan=national
    1.93 +;
    1.94 +; PRI callerid prefixes based on the given TON/NPI (dialplan)
    1.95 +; This is especially needed for euroisdn E1-PRIs
    1.96 +; 
    1.97 +; sample 1 for Germany 
    1.98 +;internationalprefix = 00
    1.99 +;nationalprefix = 0
   1.100 +;localprefix = 0711
   1.101 +;privateprefix = 07115678
   1.102 +;unknownprefix = 
   1.103 +;
   1.104 +; sample 2 for Germany 
   1.105 +;internationalprefix = +
   1.106 +;nationalprefix = +49
   1.107 +;localprefix = +49711
   1.108 +;privateprefix = +497115678
   1.109 +;unknownprefix = 
   1.110 +;
   1.111 +; PRI resetinterval: sets the time in seconds between restart of unused
   1.112 +; channels, defaults to 3600; minimum 60 seconds.  Some PBXs don't like
   1.113 +; channel restarts. so set the interval to a very long interval e.g. 100000000
   1.114 +; or 'never' to disable *entirely*.
   1.115 +;
   1.116 +;resetinterval = 3600 
   1.117 +;
   1.118 +; Overlap dialing mode (sending overlap digits)
   1.119 +;
   1.120 +;overlapdial=yes
   1.121 +;
   1.122 +; PRI Out of band indications.
   1.123 +; Enable this to report Busy and Congestion on a PRI using out-of-band
   1.124 +; notification. Inband indication, as used by Asterisk doesn't seem to work
   1.125 +; with all telcos.
   1.126 +; 
   1.127 +; outofband:      Signal Busy/Congestion out of band with RELEASE/DISCONNECT
   1.128 +; inband:         Signal Busy/Congestion using in-band tones
   1.129 +;
   1.130 +; priindication = outofband
   1.131 +;
   1.132 +; If you need to override the existing channels selection routine and force all
   1.133 +; PRI channels to be marked as exclusively selected, set this to yes.
   1.134 +; priexclusive = yes
   1.135 +;
   1.136 +; ISDN Timers
   1.137 +; All of the ISDN timers and counters that are used are configurable.  Specify
   1.138 +; the timer name, and its value (in ms for timers).
   1.139 +; K:    Layer 2 max number of outstanding unacknowledged I frames (default 7)
   1.140 +; N200: Layer 2 max number of retransmissions of a frame (default 3)
   1.141 +; T200: Layer 2 max time before retransmission of a frame (default 1000 ms)
   1.142 +; T203: Layer 2 max time without frames being exchanged (default 10000 ms)
   1.143 +; T305: Wait for DISCONNECT acknowledge (default 30000 ms)
   1.144 +; T308: Wait for RELEASE acknowledge (default 4000 ms)
   1.145 +; T309: Maintain active calls on Layer 2 disconnection (default -1, Asterisk clears calls)
   1.146 +;       EuroISDN: 6000 to 12000 ms, according to (N200 + 1) x T200 + 2s
   1.147 +;       May vary in other ISDN standards (Q.931 1993 : 90000 ms)
   1.148 +; T313: Wait for CONNECT acknowledge, CPE side only (default 3000 ms)
   1.149 +;
   1.150 +; pritimer => t200,1000
   1.151 +; pritimer => t313,4000
   1.152 +;
   1.153 +; To enable transmission of facility-based ISDN supplementary services (such
   1.154 +; as caller name from CPE over facility), enable this option.
   1.155 +; facilityenable = yes
   1.156 +;
   1.157 +;
   1.158 +; Signalling method (default is fxs).  Valid values:
   1.159 +; em:             E & M
   1.160 +; em_w:           E & M Wink
   1.161 +; featd:          Feature Group D (The fake, Adtran style, DTMF)
   1.162 +; featdmf:        Feature Group D (The real thing, MF (domestic, US))
   1.163 +; featdmf_ta:     Feature Group D (The real thing, MF (domestic, US)) through
   1.164 +;                 a Tandem Access point
   1.165 +; featb:          Feature Group B (MF (domestic, US))
   1.166 +; fgccama	  Feature Group C-CAMA (DP DNIS, MF ANI)
   1.167 +; fgccamamf	  Feature Group C-CAMA MF (MF DNIS, MF ANI)
   1.168 +; fxs_ls:         FXS (Loop Start)
   1.169 +; fxs_gs:         FXS (Ground Start)
   1.170 +; fxs_ks:         FXS (Kewl Start)
   1.171 +; fxo_ls:         FXO (Loop Start)
   1.172 +; fxo_gs:         FXO (Ground Start)
   1.173 +; fxo_ks:         FXO (Kewl Start)
   1.174 +; pri_cpe:        PRI signalling, CPE side
   1.175 +; pri_net:        PRI signalling, Network side
   1.176 +; gr303fxoks_net: GR-303 Signalling, FXO Loopstart, Network side
   1.177 +; gr303fxsks_cpe: GR-303 Signalling, FXS Loopstart, CPE side
   1.178 +; sf:	          SF (Inband Tone) Signalling
   1.179 +; sf_w:	          SF Wink
   1.180 +; sf_featd:       SF Feature Group D (The fake, Adtran style, DTMF)
   1.181 +; sf_featdmf:     SF Feature Group D (The real thing, MF (domestic, US))
   1.182 +; sf_featb:       SF Feature Group B (MF (domestic, US))
   1.183 +; e911:           E911 (MF) style signalling
   1.184 +;
   1.185 +; The following are used for Radio interfaces:
   1.186 +; fxs_rx:         Receive audio/COR on an FXS kewlstart interface (FXO at the
   1.187 +;                 channel bank)
   1.188 +; fxs_tx:         Transmit audio/PTT on an FXS loopstart interface (FXO at the
   1.189 +;                 channel bank)
   1.190 +; fxo_rx:         Receive audio/COR on an FXO loopstart interface (FXS at the
   1.191 +;                 channel bank)
   1.192 +; fxo_tx:         Transmit audio/PTT on an FXO groundstart interface (FXS at
   1.193 +;                 the channel bank)
   1.194 +; em_rx:          Receive audio/COR on an E&M interface (1-way)
   1.195 +; em_tx:          Transmit audio/PTT on an E&M interface (1-way)
   1.196 +; em_txrx:        Receive audio/COR AND Transmit audio/PTT on an E&M interface
   1.197 +;                 (2-way)
   1.198 +; em_rxtx:        Same as em_txrx (for our dyslexic friends)
   1.199 +; sf_rx:          Receive audio/COR on an SF interface (1-way)
   1.200 +; sf_tx:          Transmit audio/PTT on an SF interface (1-way)
   1.201 +; sf_txrx:        Receive audio/COR AND Transmit audio/PTT on an SF interface
   1.202 +;                 (2-way)
   1.203 +; sf_rxtx:        Same as sf_txrx (for our dyslexic friends)
   1.204 +;
   1.205 +;signalling=fxo_ls
   1.206 +;
   1.207 +; If you have an outbound signalling format that is different from format
   1.208 +; specified above (but compatible), you can specify outbound signalling format,
   1.209 +; (see below). The 'signalling' format specified will be the inbound signalling
   1.210 +; format. If you only specify 'signalling', then it will be the format for
   1.211 +; both inbound and outbound.
   1.212 +; 
   1.213 +; signalling=featdmf
   1.214 +; outsignalling=featb
   1.215 +;
   1.216 +; For Feature Group D Tandem access, to set the default CIC and OZZ use these
   1.217 +; parameters:
   1.218 +;defaultozz=0000
   1.219 +;defaultcic=303
   1.220 +;
   1.221 +; A variety of timing parameters can be specified as well
   1.222 +; Including:
   1.223 +;    prewink:     Pre-wink time (default 50ms)
   1.224 +;    preflash:    Pre-flash time (default 50ms)
   1.225 +;    wink:        Wink time (default 150ms)
   1.226 +;    flash:       Flash time (default 750ms)
   1.227 +;    start:       Start time (default 1500ms)
   1.228 +;    rxwink:      Receiver wink time (default 300ms)
   1.229 +;    rxflash:     Receiver flashtime (default 1250ms)
   1.230 +;    debounce:    Debounce timing (default 600ms)
   1.231 +;
   1.232 +;rxwink=300		; Atlas seems to use long (250ms) winks
   1.233 +;
   1.234 +; How long generated tones (DTMF and MF) will be played on the channel
   1.235 +; (in milliseconds)
   1.236 +;toneduration=100
   1.237 +;
   1.238 +; Whether or not to do distinctive ring detection on FXO lines
   1.239 +;
   1.240 +;usedistinctiveringdetection=yes
   1.241 +;distinctiveringaftercid=yes	; enable dring detection after callerid for those countries like Australia
   1.242 +				; where the ring cadence is changed *after* the callerid spill.
   1.243 +;
   1.244 +; Whether or not to use caller ID
   1.245 +;
   1.246 +;usecallerid=yes
   1.247 +;
   1.248 +; Type of caller ID signalling in use
   1.249 +;     bell     = bell202 as used in US
   1.250 +;     v23      = v23 as used in the UK
   1.251 +;     v23_jp   = v23 as used in Japan
   1.252 +;     dtmf     = DTMF as used in Denmark, Sweden and Netherlands
   1.253 +;     smdi     = Use SMDI for callerid.  Requires SMDI to be enabled (usesmdi).
   1.254 +;
   1.255 +;cidsignalling=bell
   1.256 +;
   1.257 +; What signals the start of caller ID
   1.258 +;     ring     = a ring signals the start
   1.259 +;     polarity = polarity reversal signals the start
   1.260 +;
   1.261 +;cidstart=ring
   1.262 +;
   1.263 +; Whether or not to hide outgoing caller ID (Override with *67 or *82)
   1.264 +;
   1.265 +;hidecallerid=no
   1.266 +;
   1.267 +; Whether or not to enable call waiting on internal extensions
   1.268 +; With this set to 'yes', busy extensions will hear the call-waiting
   1.269 +; tone, and can use hook-flash to switch between callers. The Dial()
   1.270 +; app will not return the "BUSY" result for extensions.
   1.271 +;
   1.272 +;callwaiting=yes
   1.273 +;
   1.274 +; Whether or not restrict outgoing caller ID (will be sent as ANI only, not
   1.275 +; available for the user)
   1.276 +; Mostly use with FXS ports
   1.277 +;
   1.278 +;restrictcid=no
   1.279 +;
   1.280 +; Whether or not use the caller ID presentation for the outgoing call that the
   1.281 +; calling switch is sending.
   1.282 +; See README.callingpres
   1.283 +;
   1.284 +;usecallingpres=yes
   1.285 +;
   1.286 +; Some countries (UK) have ring tones with different ring tones (ring-ring),
   1.287 +; which means the callerid needs to be set later on, and not just after
   1.288 +; the first ring, as per the default. 
   1.289 +;
   1.290 +;sendcalleridafter=1
   1.291 +;
   1.292 +;
   1.293 +; Support Caller*ID on Call Waiting
   1.294 +;
   1.295 +;callwaitingcallerid=yes
   1.296 +;
   1.297 +; Support three-way calling
   1.298 +;
   1.299 +;threewaycalling=yes
   1.300 +;
   1.301 +; Support flash-hook call transfer (requires three way calling)
   1.302 +; Also enables call parking (overrides the 'canpark' parameter)
   1.303 +;
   1.304 +;transfer=yes
   1.305 +;
   1.306 +; Allow call parking
   1.307 +; ('canpark=no' is overridden by 'transfer=yes')
   1.308 +;
   1.309 +;canpark=yes
   1.310 +;
   1.311 +; Support call forward variable
   1.312 +;
   1.313 +;cancallforward=yes
   1.314 +;
   1.315 +; Whether or not to support Call Return (*69)
   1.316 +;
   1.317 +;callreturn=yes
   1.318 +;
   1.319 +; Stutter dialtone support: If a mailbox is specified without a voicemail 
   1.320 +; context, then when voicemail is received in a mailbox in the default 
   1.321 +; voicemail context in voicemail.conf, taking the phone off hook will cause a
   1.322 +; stutter dialtone instead of a normal one. 
   1.323 +;
   1.324 +; If a mailbox is specified *with* a voicemail context, the same will result
   1.325 +; if voicemail received in mailbox in the specified voicemail context.
   1.326 +;
   1.327 +; for default voicemail context, the example below is fine:
   1.328 +;
   1.329 +;mailbox=1234
   1.330 +;
   1.331 +; for any other voicemail context, the following will produce the stutter tone:
   1.332 +;
   1.333 +;mailbox=1234@context 
   1.334 +;
   1.335 +; Enable echo cancellation 
   1.336 +; Use either "yes", "no", or a power of two from 32 to 256 if you wish to
   1.337 +; actually set the number of taps of cancellation.
   1.338 +;
   1.339 +; Note that when setting the number of taps, the number 256 does not translate
   1.340 +; to 256 ms of echo cancellation.  echocancel=256 means 256 / 8 = 32 ms.
   1.341 +;
   1.342 +; Note that if any of your Zaptel cards have hardware echo cancellers,
   1.343 +; then this setting only turns them on and off; numeric settings will
   1.344 +; be treated as "yes". There are no special settings required for
   1.345 +; hardware echo cancellers; when present and enabled in their kernel
   1.346 +; modules, they take precedence over the software echo canceller compiled
   1.347 +; into Zaptel automatically.
   1.348 +;
   1.349 +;echocancel=yes
   1.350 +;
   1.351 +; Generally, it is not necessary (and in fact undesirable) to echo cancel when
   1.352 +; the circuit path is entirely TDM.  You may, however, change this behavior
   1.353 +; by enabling the echo cancel during pure TDM bridging below.
   1.354 +;
   1.355 +;echocancelwhenbridged=yes
   1.356 +;
   1.357 +; In some cases, the echo canceller doesn't train quickly enough and there
   1.358 +; is echo at the beginning of the call.  Enabling echo training will cause
   1.359 +; asterisk to briefly mute the channel, send an impulse, and use the impulse
   1.360 +; response to pre-train the echo canceller so it can start out with a much
   1.361 +; closer idea of the actual echo.  Value may be "yes", "no", or a number of
   1.362 +; milliseconds to delay before training (default = 400)
   1.363 +;
   1.364 +; WARNING:  In some cases this option can make echo worse!  If you are
   1.365 +; trying to debug an echo problem, it is worth checking to see if your echo
   1.366 +; is better with the option set to yes or no.  Use whatever setting gives
   1.367 +; the best results.
   1.368 +;
   1.369 +; Note that these parameters do not apply to hardware echo cancellers.
   1.370 +;
   1.371 +;echotraining=yes
   1.372 +;echotraining=800
   1.373 +;
   1.374 +; If you are having trouble with DTMF detection, you can relax the DTMF
   1.375 +; detection parameters.  Relaxing them may make the DTMF detector more likely
   1.376 +; to have "talkoff" where DTMF is detected when it shouldn't be.
   1.377 +;
   1.378 +;relaxdtmf=yes
   1.379 +;
   1.380 +; You may also set the default receive and transmit gains (in dB)
   1.381 +;
   1.382 +;rxgain=0.0
   1.383 +;txgain=0.0
   1.384 +;
   1.385 +; Logical groups can be assigned to allow outgoing rollover.  Groups range
   1.386 +; from 0 to 63, and multiple groups can be specified.
   1.387 +;
   1.388 +;group=1
   1.389 +;
   1.390 +; Ring groups (a.k.a. call groups) and pickup groups.  If a phone is ringing
   1.391 +; and it is a member of a group which is one of your pickup groups, then
   1.392 +; you can answer it by picking up and dialling *8#.  For simple offices, just
   1.393 +; make these both the same.  Groups range from 0 to 63.
   1.394 +;
   1.395 +;callgroup=1
   1.396 +;pickupgroup=1
   1.397 +
   1.398 +;
   1.399 +; Specify whether the channel should be answered immediately or if the simple
   1.400 +; switch should provide dialtone, read digits, etc.
   1.401 +; Note: If immediate=yes the dialplan execution will always start at extension
   1.402 +; 's' priority 1 regardless of the dialed number!
   1.403 +;
   1.404 +;immediate=no
   1.405 +;
   1.406 +; Specify whether flash-hook transfers to 'busy' channels should complete or
   1.407 +; return to the caller performing the transfer (default is yes).
   1.408 +;
   1.409 +;transfertobusy=no
   1.410 +;
   1.411 +; CallerID can be set to "asreceived" or a specific number if you want to
   1.412 +; override it.  Note that "asreceived" only applies to trunk interfaces.
   1.413 +;
   1.414 +;callerid=2564286000
   1.415 +;
   1.416 +; AMA flags affects the recording of Call Detail Records.  If specified
   1.417 +; it may be 'default', 'omit', 'billing', or 'documentation'.
   1.418 +;
   1.419 +;amaflags=default
   1.420 +;
   1.421 +; Channels may be associated with an account code to ease
   1.422 +; billing
   1.423 +;
   1.424 +;accountcode=lss0101
   1.425 +;
   1.426 +; ADSI (Analog Display Services Interface) can be enabled on a per-channel
   1.427 +; basis if you have (or may have) ADSI compatible CPE equipment
   1.428 +;
   1.429 +;adsi=yes
   1.430 +;
   1.431 +; SMDI (Simplified Message Desk Interface) can be enabled on a per-channel
   1.432 +; basis if you would like that channel to behave like an SMDI message desk.
   1.433 +; The SMDI port specified should have already been defined in smdi.conf.  The
   1.434 +; default port is /dev/ttyS0.
   1.435 +;
   1.436 +;usesmdi=yes
   1.437 +;smdiport=/dev/ttyS0
   1.438 +;
   1.439 +; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D
   1.440 +; etc, it can be useful to perform busy detection either in an effort to 
   1.441 +; detect hangup or for detecting busies.  This enables listening for
   1.442 +; the beep-beep busy pattern.
   1.443 +;
   1.444 +;busydetect=yes
   1.445 +;
   1.446 +; If busydetect is enabled, it is also possible to specify how many busy tones
   1.447 +; to wait for before hanging up.  The default is 4, but better results can be
   1.448 +; achieved if set to 6 or even 8.  Mind that the higher the number, the more
   1.449 +; time that will be needed to hangup a channel, but lowers the probability
   1.450 +; that you will get random hangups.
   1.451 +;
   1.452 +;busycount=4
   1.453 +;
   1.454 +; If busydetect is enabled, it is also possible to specify the cadence of your
   1.455 +; busy signal.  In many countries, it is 500msec on, 500msec off.  Without
   1.456 +; busypattern specified, we'll accept any regular sound-silence pattern that
   1.457 +; repeats <busycount> times as a busy signal.  If you specify busypattern,
   1.458 +; then we'll further check the length of the sound (tone) and silence, which
   1.459 +; will further reduce the chance of a false positive.
   1.460 +;
   1.461 +;busypattern=500,500
   1.462 +;
   1.463 +; NOTE: In the Asterisk Makefile you'll find further options to tweak the busy
   1.464 +; detector.  If your country has a busy tone with the same length tone and
   1.465 +; silence (as many countries do), consider defining the
   1.466 +; -DBUSYDETECT_COMPARE_TONE_AND_SILENCE option.
   1.467 +;
   1.468 +; Use a polarity reversal to mark when a outgoing call is answered by the
   1.469 +; remote party.
   1.470 +;
   1.471 +;answeronpolarityswitch=yes
   1.472 +;
   1.473 +; In some countries, a polarity reversal is used to signal the disconnect of a
   1.474 +; phone line.  If the hanguponpolarityswitch option is selected, the call will
   1.475 +; be considered "hung up" on a polarity reversal.
   1.476 +;
   1.477 +;hanguponpolarityswitch=yes
   1.478 +;
   1.479 +; On trunk interfaces (FXS) it can be useful to attempt to follow the progress
   1.480 +; of a call through RINGING, BUSY, and ANSWERING.   If turned on, call
   1.481 +; progress attempts to determine answer, busy, and ringing on phone lines.
   1.482 +; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers,
   1.483 +; so don't count on it being very accurate.
   1.484 +;
   1.485 +; Few zones are supported at the time of this writing, but may be selected
   1.486 +; with "progzone"
   1.487 +;
   1.488 +; This feature can also easily detect false hangups. The symptoms of this is
   1.489 +; being disconnected in the middle of a call for no reason.
   1.490 +;
   1.491 +;callprogress=yes
   1.492 +;progzone=us
   1.493 +;
   1.494 +; FXO (FXS signalled) devices must have a timeout to determine if there was a
   1.495 +; hangup before the line was answered.  This value can be tweaked to shorten
   1.496 +; how long it takes before Zap considers a non-ringing line to have hungup.
   1.497 +;
   1.498 +;ringtimeout=8000
   1.499 +;
   1.500 +; For FXO (FXS signalled) devices, whether to use pulse dial instead of DTMF
   1.501 +;
   1.502 +;pulsedial=yes
   1.503 +;
   1.504 +; For fax detection, uncomment one of the following lines.  The default is *OFF*
   1.505 +;
   1.506 +;faxdetect=both
   1.507 +;faxdetect=incoming
   1.508 +;faxdetect=outgoing
   1.509 +;faxdetect=no
   1.510 +;
   1.511 +; This option specifies a preference for which music on hold class this channel
   1.512 +; should listen to when put on hold if the music class has not been set on the
   1.513 +; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
   1.514 +; channel putting this one on hold did not suggest a music class.
   1.515 +;
   1.516 +; If this option is set to "passthrough", then the hold message will always be
   1.517 +; passed through as signalling instead of generating hold music locally. This
   1.518 +; setting is only valid when used on a channel that uses digital signalling.
   1.519 +;
   1.520 +; This option may be specified globally, or on a per-user or per-peer basis.
   1.521 +;
   1.522 +;mohinterpret=default
   1.523 +;
   1.524 +; This option specifies which music on hold class to suggest to the peer channel
   1.525 +; when this channel places the peer on hold. It may be specified globally or on
   1.526 +; a per-user or per-peer basis.
   1.527 +;
   1.528 +;mohsuggest=default
   1.529 +;
   1.530 +; PRI channels can have an idle extension and a minunused number.  So long as
   1.531 +; at least "minunused" channels are idle, chan_zap will try to call "idledial"
   1.532 +; on them, and then dump them into the PBX in the "idleext" extension (which
   1.533 +; is of the form exten@context).  When channels are needed the "idle" calls
   1.534 +; are disconnected (so long as there are at least "minidle" calls still
   1.535 +; running, of course) to make more channels available.  The primary use of
   1.536 +; this is to create a dynamic service, where idle channels are bundled through
   1.537 +; multilink PPP, thus more efficiently utilizing combined voice/data services
   1.538 +; than conventional fixed mappings/muxings.
   1.539 +;
   1.540 +;idledial=6999
   1.541 +;idleext=6999@dialout
   1.542 +;minunused=2
   1.543 +;minidle=1
   1.544 +;
   1.545 +; Configure jitter buffers in zapata (each one is 20ms, default is 4)
   1.546 +;
   1.547 +;jitterbuffers=4
   1.548 +;
   1.549 +;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
   1.550 +; jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of a
   1.551 +                              ; ZAP channel. Defaults to "no". An enabled jitterbuffer will
   1.552 +                              ; be used only if the sending side can create and the receiving
   1.553 +                              ; side can not accept jitter. The ZAP channel can't accept jitter,
   1.554 +                              ; thus an enabled jitterbuffer on the receive ZAP side will always
   1.555 +                              ; be used if the sending side can create jitter.
   1.556 +
   1.557 +; jbmaxsize = 200             ; Max length of the jitterbuffer in milliseconds.
   1.558 +
   1.559 +; jbresyncthreshold = 1000    ; Jump in the frame timestamps over which the jitterbuffer is
   1.560 +                              ; resynchronized. Useful to improve the quality of the voice, with
   1.561 +                              ; big jumps in/broken timestamps, usually sent from exotic devices
   1.562 +                              ; and programs. Defaults to 1000.
   1.563 +
   1.564 +; jbimpl = fixed              ; Jitterbuffer implementation, used on the receiving side of a ZAP
   1.565 +                              ; channel. Two implementations are currently available - "fixed"
   1.566 +                              ; (with size always equals to jbmax-size) and "adaptive" (with
   1.567 +                              ; variable size, actually the new jb of IAX2). Defaults to fixed.
   1.568 +
   1.569 +; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
   1.570 +;-----------------------------------------------------------------------------------
   1.571 +;
   1.572 +; You can define your own custom ring cadences here.  You can define up to 8
   1.573 +; pairs.  If the silence is negative, it indicates where the callerid spill is
   1.574 +; to be placed.  Also, if you define any custom cadences, the default cadences
   1.575 +; will be turned off.
   1.576 +;
   1.577 +; Syntax is:  cadence=ring,silence[,ring,silence[...]]
   1.578 +;
   1.579 +; These are the default cadences:
   1.580 +;
   1.581 +;cadence=125,125,2000,-4000
   1.582 +;cadence=250,250,500,1000,250,250,500,-4000
   1.583 +;cadence=125,125,125,125,125,-4000
   1.584 +;cadence=1000,500,2500,-5000
   1.585 +;
   1.586 +; Each channel consists of the channel number or range.  It inherits the
   1.587 +; parameters that were specified above its declaration.
   1.588 +;
   1.589 +; For GR-303, CRV's are created like channels except they must start with the
   1.590 +; trunk group followed by a colon, e.g.: 
   1.591 +;
   1.592 +; crv => 1:1
   1.593 +; crv => 2:1-2,5-8
   1.594 +;
   1.595 +;
   1.596 +;callerid="Green Phone"<(256) 428-6121>
   1.597 +;channel => 1
   1.598 +;callerid="Black Phone"<(256) 428-6122>
   1.599 +;channel => 2
   1.600 +;callerid="CallerID Phone" <(256) 428-6123>
   1.601 +;callerid="CallerID Phone" <(630) 372-1564>
   1.602 +;callerid="CallerID Phone" <(256) 704-4666>
   1.603 +;channel => 3
   1.604 +;callerid="Pac Tel Phone" <(256) 428-6124>
   1.605 +;channel => 4
   1.606 +;callerid="Uniden Dead" <(256) 428-6125>
   1.607 +;channel => 5
   1.608 +;callerid="Cortelco 2500" <(256) 428-6126>
   1.609 +;channel => 6
   1.610 +;callerid="Main TA 750" <(256) 428-6127>
   1.611 +;channel => 44
   1.612 +;
   1.613 +; For example, maybe we have some other channels which start out in a
   1.614 +; different context and use E & M signalling instead.
   1.615 +;
   1.616 +;context=remote
   1.617 +;sigalling=em
   1.618 +;channel => 15
   1.619 +;channel => 16
   1.620 +
   1.621 +;signalling=em_w
   1.622 +;
   1.623 +; All those in group 0 I'll use for outgoing calls
   1.624 +;
   1.625 +; Strip most significant digit (9) before sending
   1.626 +;
   1.627 +;stripmsd=1
   1.628 +;callerid=asreceived
   1.629 +;group=0
   1.630 +;signalling=fxs_ls
   1.631 +;channel => 45
   1.632 +
   1.633 +;signalling=fxo_ls
   1.634 +;group=1
   1.635 +;callerid="Joe Schmoe" <(256) 428-6131>
   1.636 +;channel => 25
   1.637 +;callerid="Megan May" <(256) 428-6132>
   1.638 +;channel => 26
   1.639 +;callerid="Suzy Queue" <(256) 428-6233>
   1.640 +;channel => 27
   1.641 +;callerid="Larry Moe" <(256) 428-6234>
   1.642 +;channel => 28
   1.643 +;
   1.644 +; Sample PRI (CPE) config:  Specify the switchtype, the signalling as either
   1.645 +; pri_cpe or pri_net for CPE or Network termination, and generally you will
   1.646 +; want to create a single "group" for all channels of the PRI.
   1.647 +;
   1.648 +; switchtype = national
   1.649 +; signalling = pri_cpe
   1.650 +; group = 2
   1.651 +; channel => 1-23
   1.652 +
   1.653 +;
   1.654 +
   1.655 +;  Used for distinctive ring support for x100p.
   1.656 +;  You can see the dringX patterns is to set any one of the dringXcontext fields
   1.657 +;  and they will be printed on the console when an inbound call comes in.
   1.658 +;
   1.659 +;dring1=95,0,0 
   1.660 +;dring1context=internal1 
   1.661 +;dring2=325,95,0 
   1.662 +;dring2context=internal2 
   1.663 +; If no pattern is matched here is where we go.
   1.664 +;context=default
   1.665 +;channel => 1 
   1.666 +

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