asterisk/zapata.conf

Mon, 27 Apr 2009 12:19:05 +0200

author
Michael Schloh von Bennewitz <michael@schloh.com>
date
Mon, 27 Apr 2009 12:19:05 +0200
changeset 202
f29abea29121
permissions
-rw-r--r--

Import new package spec for introduction into repository.

     1 ;
     2 ; Zapata telephony interface
     3 ;
     4 ; Configuration file
     5 ;
     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.
    14 ;[trunkgroups]
    15 ;
    16 ; Trunk groups are used for NFAS or GR-303 connections.
    17 ;
    18 ; Group: Defines a trunk group.  
    19 ;        trunkgroup => <trunkgroup>,<dchannel>[,<backup1>...]
    20 ;
    21 ;        trunkgroup  is the numerical trunk group to create
    22 ;        dchannel    is the zap channel which will have the 
    23 ;                    d-channel for the trunk.
    24 ;        backup1     is an optional list of backup d-channels.
    25 ;
    26 ;trunkgroup => 1,24,48
    27 ;trunkgroup => 1,24
    28 ;
    29 ; Spanmap: Associates a span with a trunk group
    30 ;        spanmap => <zapspan>,<trunkgroup>[,<logicalspan>]
    31 ;
    32 ;        zapspan     is the zap span number to associate
    33 ;        trunkgroup  is the trunkgroup (specified above) for the mapping
    34 ;        logicalspan is the logical span number within the trunk group to use.
    35 ;                    if unspecified, no logical span number is used.
    36 ;
    37 ;spanmap => 1,1,1
    38 ;spanmap => 2,1,2
    39 ;spanmap => 3,1,3
    40 ;spanmap => 4,1,4
    42 ;[channels]
    43 ;
    44 ; Default language
    45 ;
    46 ;language=en
    47 ;
    48 ; Default context
    49 ;
    50 ;context=default
    51 ;
    52 ; Switchtype:  Only used for PRI.
    53 ;
    54 ; national:	  National ISDN 2 (default)
    55 ; dms100:	  Nortel DMS100
    56 ; 4ess:           AT&T 4ESS
    57 ; 5ess:	          Lucent 5ESS
    58 ; euroisdn:       EuroISDN
    59 ; ni1:            Old National ISDN 1
    60 ; qsig:           Q.SIG
    61 ;
    62 ;switchtype=national
    63 ;
    64 ; Some switches (AT&T especially) require network specific facility IE
    65 ; supported values are currently 'none', 'sdn', 'megacom', 'tollfreemegacom', 'accunet'
    66 ;
    67 ;nsf=none
    68 ;
    69 ; PRI Dialplan:  Only RARELY used for PRI.
    70 ;
    71 ; unknown:        Unknown
    72 ; private:        Private ISDN
    73 ; local:          Local ISDN
    74 ; national:	  National ISDN
    75 ; international:  International ISDN
    76 ; dynamic:	  Dynamically selects the appropriate dialplan
    77 ;
    78 ;pridialplan=national
    79 ;
    80 ; PRI Local Dialplan:  Only RARELY used for PRI (sets the calling number's numbering plan)
    81 ;
    82 ; unknown:        Unknown
    83 ; private:        Private ISDN
    84 ; local:          Local ISDN
    85 ; national:	  National ISDN
    86 ; international:  International ISDN
    87 ; dynamic:	  Dynamically selects the appropriate dialplan
    88 ;
    89 ;prilocaldialplan=national
    90 ;
    91 ; PRI callerid prefixes based on the given TON/NPI (dialplan)
    92 ; This is especially needed for euroisdn E1-PRIs
    93 ; 
    94 ; sample 1 for Germany 
    95 ;internationalprefix = 00
    96 ;nationalprefix = 0
    97 ;localprefix = 0711
    98 ;privateprefix = 07115678
    99 ;unknownprefix = 
   100 ;
   101 ; sample 2 for Germany 
   102 ;internationalprefix = +
   103 ;nationalprefix = +49
   104 ;localprefix = +49711
   105 ;privateprefix = +497115678
   106 ;unknownprefix = 
   107 ;
   108 ; PRI resetinterval: sets the time in seconds between restart of unused
   109 ; channels, defaults to 3600; minimum 60 seconds.  Some PBXs don't like
   110 ; channel restarts. so set the interval to a very long interval e.g. 100000000
   111 ; or 'never' to disable *entirely*.
   112 ;
   113 ;resetinterval = 3600 
   114 ;
   115 ; Overlap dialing mode (sending overlap digits)
   116 ;
   117 ;overlapdial=yes
   118 ;
   119 ; PRI Out of band indications.
   120 ; Enable this to report Busy and Congestion on a PRI using out-of-band
   121 ; notification. Inband indication, as used by Asterisk doesn't seem to work
   122 ; with all telcos.
   123 ; 
   124 ; outofband:      Signal Busy/Congestion out of band with RELEASE/DISCONNECT
   125 ; inband:         Signal Busy/Congestion using in-band tones
   126 ;
   127 ; priindication = outofband
   128 ;
   129 ; If you need to override the existing channels selection routine and force all
   130 ; PRI channels to be marked as exclusively selected, set this to yes.
   131 ; priexclusive = yes
   132 ;
   133 ; ISDN Timers
   134 ; All of the ISDN timers and counters that are used are configurable.  Specify
   135 ; the timer name, and its value (in ms for timers).
   136 ; K:    Layer 2 max number of outstanding unacknowledged I frames (default 7)
   137 ; N200: Layer 2 max number of retransmissions of a frame (default 3)
   138 ; T200: Layer 2 max time before retransmission of a frame (default 1000 ms)
   139 ; T203: Layer 2 max time without frames being exchanged (default 10000 ms)
   140 ; T305: Wait for DISCONNECT acknowledge (default 30000 ms)
   141 ; T308: Wait for RELEASE acknowledge (default 4000 ms)
   142 ; T309: Maintain active calls on Layer 2 disconnection (default -1, Asterisk clears calls)
   143 ;       EuroISDN: 6000 to 12000 ms, according to (N200 + 1) x T200 + 2s
   144 ;       May vary in other ISDN standards (Q.931 1993 : 90000 ms)
   145 ; T313: Wait for CONNECT acknowledge, CPE side only (default 3000 ms)
   146 ;
   147 ; pritimer => t200,1000
   148 ; pritimer => t313,4000
   149 ;
   150 ; To enable transmission of facility-based ISDN supplementary services (such
   151 ; as caller name from CPE over facility), enable this option.
   152 ; facilityenable = yes
   153 ;
   154 ;
   155 ; Signalling method (default is fxs).  Valid values:
   156 ; em:             E & M
   157 ; em_w:           E & M Wink
   158 ; featd:          Feature Group D (The fake, Adtran style, DTMF)
   159 ; featdmf:        Feature Group D (The real thing, MF (domestic, US))
   160 ; featdmf_ta:     Feature Group D (The real thing, MF (domestic, US)) through
   161 ;                 a Tandem Access point
   162 ; featb:          Feature Group B (MF (domestic, US))
   163 ; fgccama	  Feature Group C-CAMA (DP DNIS, MF ANI)
   164 ; fgccamamf	  Feature Group C-CAMA MF (MF DNIS, MF ANI)
   165 ; fxs_ls:         FXS (Loop Start)
   166 ; fxs_gs:         FXS (Ground Start)
   167 ; fxs_ks:         FXS (Kewl Start)
   168 ; fxo_ls:         FXO (Loop Start)
   169 ; fxo_gs:         FXO (Ground Start)
   170 ; fxo_ks:         FXO (Kewl Start)
   171 ; pri_cpe:        PRI signalling, CPE side
   172 ; pri_net:        PRI signalling, Network side
   173 ; gr303fxoks_net: GR-303 Signalling, FXO Loopstart, Network side
   174 ; gr303fxsks_cpe: GR-303 Signalling, FXS Loopstart, CPE side
   175 ; sf:	          SF (Inband Tone) Signalling
   176 ; sf_w:	          SF Wink
   177 ; sf_featd:       SF Feature Group D (The fake, Adtran style, DTMF)
   178 ; sf_featdmf:     SF Feature Group D (The real thing, MF (domestic, US))
   179 ; sf_featb:       SF Feature Group B (MF (domestic, US))
   180 ; e911:           E911 (MF) style signalling
   181 ;
   182 ; The following are used for Radio interfaces:
   183 ; fxs_rx:         Receive audio/COR on an FXS kewlstart interface (FXO at the
   184 ;                 channel bank)
   185 ; fxs_tx:         Transmit audio/PTT on an FXS loopstart interface (FXO at the
   186 ;                 channel bank)
   187 ; fxo_rx:         Receive audio/COR on an FXO loopstart interface (FXS at the
   188 ;                 channel bank)
   189 ; fxo_tx:         Transmit audio/PTT on an FXO groundstart interface (FXS at
   190 ;                 the channel bank)
   191 ; em_rx:          Receive audio/COR on an E&M interface (1-way)
   192 ; em_tx:          Transmit audio/PTT on an E&M interface (1-way)
   193 ; em_txrx:        Receive audio/COR AND Transmit audio/PTT on an E&M interface
   194 ;                 (2-way)
   195 ; em_rxtx:        Same as em_txrx (for our dyslexic friends)
   196 ; sf_rx:          Receive audio/COR on an SF interface (1-way)
   197 ; sf_tx:          Transmit audio/PTT on an SF interface (1-way)
   198 ; sf_txrx:        Receive audio/COR AND Transmit audio/PTT on an SF interface
   199 ;                 (2-way)
   200 ; sf_rxtx:        Same as sf_txrx (for our dyslexic friends)
   201 ;
   202 ;signalling=fxo_ls
   203 ;
   204 ; If you have an outbound signalling format that is different from format
   205 ; specified above (but compatible), you can specify outbound signalling format,
   206 ; (see below). The 'signalling' format specified will be the inbound signalling
   207 ; format. If you only specify 'signalling', then it will be the format for
   208 ; both inbound and outbound.
   209 ; 
   210 ; signalling=featdmf
   211 ; outsignalling=featb
   212 ;
   213 ; For Feature Group D Tandem access, to set the default CIC and OZZ use these
   214 ; parameters:
   215 ;defaultozz=0000
   216 ;defaultcic=303
   217 ;
   218 ; A variety of timing parameters can be specified as well
   219 ; Including:
   220 ;    prewink:     Pre-wink time (default 50ms)
   221 ;    preflash:    Pre-flash time (default 50ms)
   222 ;    wink:        Wink time (default 150ms)
   223 ;    flash:       Flash time (default 750ms)
   224 ;    start:       Start time (default 1500ms)
   225 ;    rxwink:      Receiver wink time (default 300ms)
   226 ;    rxflash:     Receiver flashtime (default 1250ms)
   227 ;    debounce:    Debounce timing (default 600ms)
   228 ;
   229 ;rxwink=300		; Atlas seems to use long (250ms) winks
   230 ;
   231 ; How long generated tones (DTMF and MF) will be played on the channel
   232 ; (in milliseconds)
   233 ;toneduration=100
   234 ;
   235 ; Whether or not to do distinctive ring detection on FXO lines
   236 ;
   237 ;usedistinctiveringdetection=yes
   238 ;distinctiveringaftercid=yes	; enable dring detection after callerid for those countries like Australia
   239 				; where the ring cadence is changed *after* the callerid spill.
   240 ;
   241 ; Whether or not to use caller ID
   242 ;
   243 ;usecallerid=yes
   244 ;
   245 ; Type of caller ID signalling in use
   246 ;     bell     = bell202 as used in US
   247 ;     v23      = v23 as used in the UK
   248 ;     v23_jp   = v23 as used in Japan
   249 ;     dtmf     = DTMF as used in Denmark, Sweden and Netherlands
   250 ;     smdi     = Use SMDI for callerid.  Requires SMDI to be enabled (usesmdi).
   251 ;
   252 ;cidsignalling=bell
   253 ;
   254 ; What signals the start of caller ID
   255 ;     ring     = a ring signals the start
   256 ;     polarity = polarity reversal signals the start
   257 ;
   258 ;cidstart=ring
   259 ;
   260 ; Whether or not to hide outgoing caller ID (Override with *67 or *82)
   261 ;
   262 ;hidecallerid=no
   263 ;
   264 ; Whether or not to enable call waiting on internal extensions
   265 ; With this set to 'yes', busy extensions will hear the call-waiting
   266 ; tone, and can use hook-flash to switch between callers. The Dial()
   267 ; app will not return the "BUSY" result for extensions.
   268 ;
   269 ;callwaiting=yes
   270 ;
   271 ; Whether or not restrict outgoing caller ID (will be sent as ANI only, not
   272 ; available for the user)
   273 ; Mostly use with FXS ports
   274 ;
   275 ;restrictcid=no
   276 ;
   277 ; Whether or not use the caller ID presentation for the outgoing call that the
   278 ; calling switch is sending.
   279 ; See README.callingpres
   280 ;
   281 ;usecallingpres=yes
   282 ;
   283 ; Some countries (UK) have ring tones with different ring tones (ring-ring),
   284 ; which means the callerid needs to be set later on, and not just after
   285 ; the first ring, as per the default. 
   286 ;
   287 ;sendcalleridafter=1
   288 ;
   289 ;
   290 ; Support Caller*ID on Call Waiting
   291 ;
   292 ;callwaitingcallerid=yes
   293 ;
   294 ; Support three-way calling
   295 ;
   296 ;threewaycalling=yes
   297 ;
   298 ; Support flash-hook call transfer (requires three way calling)
   299 ; Also enables call parking (overrides the 'canpark' parameter)
   300 ;
   301 ;transfer=yes
   302 ;
   303 ; Allow call parking
   304 ; ('canpark=no' is overridden by 'transfer=yes')
   305 ;
   306 ;canpark=yes
   307 ;
   308 ; Support call forward variable
   309 ;
   310 ;cancallforward=yes
   311 ;
   312 ; Whether or not to support Call Return (*69)
   313 ;
   314 ;callreturn=yes
   315 ;
   316 ; Stutter dialtone support: If a mailbox is specified without a voicemail 
   317 ; context, then when voicemail is received in a mailbox in the default 
   318 ; voicemail context in voicemail.conf, taking the phone off hook will cause a
   319 ; stutter dialtone instead of a normal one. 
   320 ;
   321 ; If a mailbox is specified *with* a voicemail context, the same will result
   322 ; if voicemail received in mailbox in the specified voicemail context.
   323 ;
   324 ; for default voicemail context, the example below is fine:
   325 ;
   326 ;mailbox=1234
   327 ;
   328 ; for any other voicemail context, the following will produce the stutter tone:
   329 ;
   330 ;mailbox=1234@context 
   331 ;
   332 ; Enable echo cancellation 
   333 ; Use either "yes", "no", or a power of two from 32 to 256 if you wish to
   334 ; actually set the number of taps of cancellation.
   335 ;
   336 ; Note that when setting the number of taps, the number 256 does not translate
   337 ; to 256 ms of echo cancellation.  echocancel=256 means 256 / 8 = 32 ms.
   338 ;
   339 ; Note that if any of your Zaptel cards have hardware echo cancellers,
   340 ; then this setting only turns them on and off; numeric settings will
   341 ; be treated as "yes". There are no special settings required for
   342 ; hardware echo cancellers; when present and enabled in their kernel
   343 ; modules, they take precedence over the software echo canceller compiled
   344 ; into Zaptel automatically.
   345 ;
   346 ;echocancel=yes
   347 ;
   348 ; Generally, it is not necessary (and in fact undesirable) to echo cancel when
   349 ; the circuit path is entirely TDM.  You may, however, change this behavior
   350 ; by enabling the echo cancel during pure TDM bridging below.
   351 ;
   352 ;echocancelwhenbridged=yes
   353 ;
   354 ; In some cases, the echo canceller doesn't train quickly enough and there
   355 ; is echo at the beginning of the call.  Enabling echo training will cause
   356 ; asterisk to briefly mute the channel, send an impulse, and use the impulse
   357 ; response to pre-train the echo canceller so it can start out with a much
   358 ; closer idea of the actual echo.  Value may be "yes", "no", or a number of
   359 ; milliseconds to delay before training (default = 400)
   360 ;
   361 ; WARNING:  In some cases this option can make echo worse!  If you are
   362 ; trying to debug an echo problem, it is worth checking to see if your echo
   363 ; is better with the option set to yes or no.  Use whatever setting gives
   364 ; the best results.
   365 ;
   366 ; Note that these parameters do not apply to hardware echo cancellers.
   367 ;
   368 ;echotraining=yes
   369 ;echotraining=800
   370 ;
   371 ; If you are having trouble with DTMF detection, you can relax the DTMF
   372 ; detection parameters.  Relaxing them may make the DTMF detector more likely
   373 ; to have "talkoff" where DTMF is detected when it shouldn't be.
   374 ;
   375 ;relaxdtmf=yes
   376 ;
   377 ; You may also set the default receive and transmit gains (in dB)
   378 ;
   379 ;rxgain=0.0
   380 ;txgain=0.0
   381 ;
   382 ; Logical groups can be assigned to allow outgoing rollover.  Groups range
   383 ; from 0 to 63, and multiple groups can be specified.
   384 ;
   385 ;group=1
   386 ;
   387 ; Ring groups (a.k.a. call groups) and pickup groups.  If a phone is ringing
   388 ; and it is a member of a group which is one of your pickup groups, then
   389 ; you can answer it by picking up and dialling *8#.  For simple offices, just
   390 ; make these both the same.  Groups range from 0 to 63.
   391 ;
   392 ;callgroup=1
   393 ;pickupgroup=1
   395 ;
   396 ; Specify whether the channel should be answered immediately or if the simple
   397 ; switch should provide dialtone, read digits, etc.
   398 ; Note: If immediate=yes the dialplan execution will always start at extension
   399 ; 's' priority 1 regardless of the dialed number!
   400 ;
   401 ;immediate=no
   402 ;
   403 ; Specify whether flash-hook transfers to 'busy' channels should complete or
   404 ; return to the caller performing the transfer (default is yes).
   405 ;
   406 ;transfertobusy=no
   407 ;
   408 ; CallerID can be set to "asreceived" or a specific number if you want to
   409 ; override it.  Note that "asreceived" only applies to trunk interfaces.
   410 ;
   411 ;callerid=2564286000
   412 ;
   413 ; AMA flags affects the recording of Call Detail Records.  If specified
   414 ; it may be 'default', 'omit', 'billing', or 'documentation'.
   415 ;
   416 ;amaflags=default
   417 ;
   418 ; Channels may be associated with an account code to ease
   419 ; billing
   420 ;
   421 ;accountcode=lss0101
   422 ;
   423 ; ADSI (Analog Display Services Interface) can be enabled on a per-channel
   424 ; basis if you have (or may have) ADSI compatible CPE equipment
   425 ;
   426 ;adsi=yes
   427 ;
   428 ; SMDI (Simplified Message Desk Interface) can be enabled on a per-channel
   429 ; basis if you would like that channel to behave like an SMDI message desk.
   430 ; The SMDI port specified should have already been defined in smdi.conf.  The
   431 ; default port is /dev/ttyS0.
   432 ;
   433 ;usesmdi=yes
   434 ;smdiport=/dev/ttyS0
   435 ;
   436 ; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D
   437 ; etc, it can be useful to perform busy detection either in an effort to 
   438 ; detect hangup or for detecting busies.  This enables listening for
   439 ; the beep-beep busy pattern.
   440 ;
   441 ;busydetect=yes
   442 ;
   443 ; If busydetect is enabled, it is also possible to specify how many busy tones
   444 ; to wait for before hanging up.  The default is 4, but better results can be
   445 ; achieved if set to 6 or even 8.  Mind that the higher the number, the more
   446 ; time that will be needed to hangup a channel, but lowers the probability
   447 ; that you will get random hangups.
   448 ;
   449 ;busycount=4
   450 ;
   451 ; If busydetect is enabled, it is also possible to specify the cadence of your
   452 ; busy signal.  In many countries, it is 500msec on, 500msec off.  Without
   453 ; busypattern specified, we'll accept any regular sound-silence pattern that
   454 ; repeats <busycount> times as a busy signal.  If you specify busypattern,
   455 ; then we'll further check the length of the sound (tone) and silence, which
   456 ; will further reduce the chance of a false positive.
   457 ;
   458 ;busypattern=500,500
   459 ;
   460 ; NOTE: In the Asterisk Makefile you'll find further options to tweak the busy
   461 ; detector.  If your country has a busy tone with the same length tone and
   462 ; silence (as many countries do), consider defining the
   463 ; -DBUSYDETECT_COMPARE_TONE_AND_SILENCE option.
   464 ;
   465 ; Use a polarity reversal to mark when a outgoing call is answered by the
   466 ; remote party.
   467 ;
   468 ;answeronpolarityswitch=yes
   469 ;
   470 ; In some countries, a polarity reversal is used to signal the disconnect of a
   471 ; phone line.  If the hanguponpolarityswitch option is selected, the call will
   472 ; be considered "hung up" on a polarity reversal.
   473 ;
   474 ;hanguponpolarityswitch=yes
   475 ;
   476 ; On trunk interfaces (FXS) it can be useful to attempt to follow the progress
   477 ; of a call through RINGING, BUSY, and ANSWERING.   If turned on, call
   478 ; progress attempts to determine answer, busy, and ringing on phone lines.
   479 ; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers,
   480 ; so don't count on it being very accurate.
   481 ;
   482 ; Few zones are supported at the time of this writing, but may be selected
   483 ; with "progzone"
   484 ;
   485 ; This feature can also easily detect false hangups. The symptoms of this is
   486 ; being disconnected in the middle of a call for no reason.
   487 ;
   488 ;callprogress=yes
   489 ;progzone=us
   490 ;
   491 ; FXO (FXS signalled) devices must have a timeout to determine if there was a
   492 ; hangup before the line was answered.  This value can be tweaked to shorten
   493 ; how long it takes before Zap considers a non-ringing line to have hungup.
   494 ;
   495 ;ringtimeout=8000
   496 ;
   497 ; For FXO (FXS signalled) devices, whether to use pulse dial instead of DTMF
   498 ;
   499 ;pulsedial=yes
   500 ;
   501 ; For fax detection, uncomment one of the following lines.  The default is *OFF*
   502 ;
   503 ;faxdetect=both
   504 ;faxdetect=incoming
   505 ;faxdetect=outgoing
   506 ;faxdetect=no
   507 ;
   508 ; This option specifies a preference for which music on hold class this channel
   509 ; should listen to when put on hold if the music class has not been set on the
   510 ; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
   511 ; channel putting this one on hold did not suggest a music class.
   512 ;
   513 ; If this option is set to "passthrough", then the hold message will always be
   514 ; passed through as signalling instead of generating hold music locally. This
   515 ; setting is only valid when used on a channel that uses digital signalling.
   516 ;
   517 ; This option may be specified globally, or on a per-user or per-peer basis.
   518 ;
   519 ;mohinterpret=default
   520 ;
   521 ; This option specifies which music on hold class to suggest to the peer channel
   522 ; when this channel places the peer on hold. It may be specified globally or on
   523 ; a per-user or per-peer basis.
   524 ;
   525 ;mohsuggest=default
   526 ;
   527 ; PRI channels can have an idle extension and a minunused number.  So long as
   528 ; at least "minunused" channels are idle, chan_zap will try to call "idledial"
   529 ; on them, and then dump them into the PBX in the "idleext" extension (which
   530 ; is of the form exten@context).  When channels are needed the "idle" calls
   531 ; are disconnected (so long as there are at least "minidle" calls still
   532 ; running, of course) to make more channels available.  The primary use of
   533 ; this is to create a dynamic service, where idle channels are bundled through
   534 ; multilink PPP, thus more efficiently utilizing combined voice/data services
   535 ; than conventional fixed mappings/muxings.
   536 ;
   537 ;idledial=6999
   538 ;idleext=6999@dialout
   539 ;minunused=2
   540 ;minidle=1
   541 ;
   542 ; Configure jitter buffers in zapata (each one is 20ms, default is 4)
   543 ;
   544 ;jitterbuffers=4
   545 ;
   546 ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
   547 ; jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of a
   548                               ; ZAP channel. Defaults to "no". An enabled jitterbuffer will
   549                               ; be used only if the sending side can create and the receiving
   550                               ; side can not accept jitter. The ZAP channel can't accept jitter,
   551                               ; thus an enabled jitterbuffer on the receive ZAP side will always
   552                               ; be used if the sending side can create jitter.
   554 ; jbmaxsize = 200             ; Max length of the jitterbuffer in milliseconds.
   556 ; jbresyncthreshold = 1000    ; Jump in the frame timestamps over which the jitterbuffer is
   557                               ; resynchronized. Useful to improve the quality of the voice, with
   558                               ; big jumps in/broken timestamps, usually sent from exotic devices
   559                               ; and programs. Defaults to 1000.
   561 ; jbimpl = fixed              ; Jitterbuffer implementation, used on the receiving side of a ZAP
   562                               ; channel. Two implementations are currently available - "fixed"
   563                               ; (with size always equals to jbmax-size) and "adaptive" (with
   564                               ; variable size, actually the new jb of IAX2). Defaults to fixed.
   566 ; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
   567 ;-----------------------------------------------------------------------------------
   568 ;
   569 ; You can define your own custom ring cadences here.  You can define up to 8
   570 ; pairs.  If the silence is negative, it indicates where the callerid spill is
   571 ; to be placed.  Also, if you define any custom cadences, the default cadences
   572 ; will be turned off.
   573 ;
   574 ; Syntax is:  cadence=ring,silence[,ring,silence[...]]
   575 ;
   576 ; These are the default cadences:
   577 ;
   578 ;cadence=125,125,2000,-4000
   579 ;cadence=250,250,500,1000,250,250,500,-4000
   580 ;cadence=125,125,125,125,125,-4000
   581 ;cadence=1000,500,2500,-5000
   582 ;
   583 ; Each channel consists of the channel number or range.  It inherits the
   584 ; parameters that were specified above its declaration.
   585 ;
   586 ; For GR-303, CRV's are created like channels except they must start with the
   587 ; trunk group followed by a colon, e.g.: 
   588 ;
   589 ; crv => 1:1
   590 ; crv => 2:1-2,5-8
   591 ;
   592 ;
   593 ;callerid="Green Phone"<(256) 428-6121>
   594 ;channel => 1
   595 ;callerid="Black Phone"<(256) 428-6122>
   596 ;channel => 2
   597 ;callerid="CallerID Phone" <(256) 428-6123>
   598 ;callerid="CallerID Phone" <(630) 372-1564>
   599 ;callerid="CallerID Phone" <(256) 704-4666>
   600 ;channel => 3
   601 ;callerid="Pac Tel Phone" <(256) 428-6124>
   602 ;channel => 4
   603 ;callerid="Uniden Dead" <(256) 428-6125>
   604 ;channel => 5
   605 ;callerid="Cortelco 2500" <(256) 428-6126>
   606 ;channel => 6
   607 ;callerid="Main TA 750" <(256) 428-6127>
   608 ;channel => 44
   609 ;
   610 ; For example, maybe we have some other channels which start out in a
   611 ; different context and use E & M signalling instead.
   612 ;
   613 ;context=remote
   614 ;sigalling=em
   615 ;channel => 15
   616 ;channel => 16
   618 ;signalling=em_w
   619 ;
   620 ; All those in group 0 I'll use for outgoing calls
   621 ;
   622 ; Strip most significant digit (9) before sending
   623 ;
   624 ;stripmsd=1
   625 ;callerid=asreceived
   626 ;group=0
   627 ;signalling=fxs_ls
   628 ;channel => 45
   630 ;signalling=fxo_ls
   631 ;group=1
   632 ;callerid="Joe Schmoe" <(256) 428-6131>
   633 ;channel => 25
   634 ;callerid="Megan May" <(256) 428-6132>
   635 ;channel => 26
   636 ;callerid="Suzy Queue" <(256) 428-6233>
   637 ;channel => 27
   638 ;callerid="Larry Moe" <(256) 428-6234>
   639 ;channel => 28
   640 ;
   641 ; Sample PRI (CPE) config:  Specify the switchtype, the signalling as either
   642 ; pri_cpe or pri_net for CPE or Network termination, and generally you will
   643 ; want to create a single "group" for all channels of the PRI.
   644 ;
   645 ; switchtype = national
   646 ; signalling = pri_cpe
   647 ; group = 2
   648 ; channel => 1-23
   650 ;
   652 ;  Used for distinctive ring support for x100p.
   653 ;  You can see the dringX patterns is to set any one of the dringXcontext fields
   654 ;  and they will be printed on the console when an inbound call comes in.
   655 ;
   656 ;dring1=95,0,0 
   657 ;dring1context=internal1 
   658 ;dring2=325,95,0 
   659 ;dring2context=internal2 
   660 ; If no pattern is matched here is where we go.
   661 ;context=default
   662 ;channel => 1 

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