asterisk/zapata.conf

Fri, 15 Oct 2010 18:46:25 +0200

author
Michael Schloh von Bennewitz <michael@schloh.com>
date
Fri, 15 Oct 2010 18:46:25 +0200
changeset 261
4f973c756446
permissions
-rw-r--r--

Update copyright, file server URL, modify doc and link logic.
Now documentation is installed by default to the correct path,
and QtCreator links against Qt shared libraries instead of Qt
static libraries. This unfortunate change supports Nokia's
unfortunate decision to poorly support static linking in Qt.

     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 

mercurial