[Asterisk-Users] Probs with outbound calls
Colin Tordale
ctordale at gmail.com
Fri Dec 23 15:04:39 MST 2005
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#
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
#
# First come the span definitions, in the format
# span=<span num>,<timing>,<line build out (LBO)>,<framing>,<coding>[,yellow]
#
# The timing parameter determines the selection of primary, secondary, and
# so on sync sources. If this span should be considered a primary sync
# source, then give it a value of "1". For a secondary, use "2", and so on.
# To not use this as a sync source, just use "0"
#
# The line build-out (or LBO) is an integer, from the following table:
# 0: 0 db (CSU) / 0-133 feet (DSX-1)
# 1: 133-266 feet (DSX-1)
# 2: 266-399 feet (DSX-1)
# 3: 399-533 feet (DSX-1)
# 4: 533-655 feet (DSX-1)
# 5: -7.5db (CSU)
# 6: -15db (CSU)
# 7: -22.5db (CSU)
#
# The framing is one of "d4" or "esf" for T1 or "cas" or "ccs" for E1
#
# Note: "d4" could be referred to as "sf" or "superframe"
#
# The coding is one of "ami" or "b8zs" for T1 or "ami" or "hdb3" for E1
#
# E1's may have the additional keyword "crc4" to enable CRC4 checking
#
# If the keyword "yellow" follows, yellow alarm is transmitted when no
# channels are open.
#
#span=1,0,0,esf,b8zs
#span=2,1,0,esf,b8zs
#span=3,0,0,ccs,hdb3,crc4
#
# Next come the dynamic span definitions, in the form:
# dynamic=<driver>,<address>,<numchans>,<timing>
#
# Where <driver> is the name of the driver (e.g. eth), <address> is the
# driver specific address (like a MAC for eth), <numchans> is the number
# of channels, and <timing> is a timing priority, like for a normal span.
# use "0" to not use this as a timing source, or prioritize them as
# primary, secondard, etc. Note that you MUST have a REAL zaptel device
# if you are not using external timing.
#
# dynamic=eth,eth0/00:02:b3:35:43:9c,24,0
#
# Next come the definitions for using the channels. The format is:
# <device>=<channel list>
#
# Valid devices are:
#
# "e&m" : Channel(s) are signalled using E&M signalling (specific
# implementation, such as Immediate, Wink, or Feature Group D
# are handled by the userspace library).
# "fxsls" : Channel(s) are signalled using FXS Loopstart protocol.
# "fxsgs" : Channel(s) are signalled using FXS Groundstart protocol.
# "fxsks" : Channel(s) are signalled using FXS Koolstart protocol.
# "fxols" : Channel(s) are signalled using FXO Loopstart protocol.
# "fxogs" : Channel(s) are signalled using FXO Groundstart protocol.
# "fxoks" : Channel(s) are signalled using FXO Koolstart protocol.
# "sf" : Channel(s) are signalled using in-band single freq tone.
# Syntax as follows:
# channel# => sf:<rxfreq>,<rxbw>,<rxflag>,<txfreq>,<txlevel>,<txflag>
# rxfreq is rx tone freq in hz, rxbw is rx notch (and decode)
# bandwith in hz (typically 10.0), rxflag is either 'normal' or
# 'inverted', txfreq is tx tone freq in hz, txlevel is tx tone
# level in dbm, txflag is either 'normal' or 'inverted'. Set
# rxfreq or txfreq to 0.0 if that tone is not desired.
# "unused" : No signalling is performed, each channel in the list remains idle
# "clear" : Channel(s) are bundled into a single span. No conversion or
# signalling is performed, and raw data is available on the master.
# "indclear": Like "clear" except all channels are treated individually and
# are not bundled. "bchan" is an alias for this.
# "rawhdlc" : The zaptel driver performs HDLC encoding and decoding on the
# bundle, and the resulting data is communicated via the master
# device.
# "fcshdlc" : The zapdel driver performs HDLC encoding and decoding on the
# bundle and also performs incoming and outgoing FCS insertion
# and verification. "dchan" is an alias for this.
# "nethdlc" : The zaptel driver bundles the channels together into an
# hdlc network device, which in turn can be configured with
# sethdlc (available separately).
# "dacs" : The zaptel driver cross connects the channels starting at
# the channel number listed at the end, after a colon
# "dacsrbs" : The zaptel driver cross connects the channels starting at
# the channel number listed at the end, after a colon and
# also performs the DACSing of RBS bits
#
# The channel list is a comma-separated list of channels or ranges, for
# example:
#
# 1,3,5 (channels one, three, and five)
# 16-23, 29 (channels 16 through 23, as well as channel 29
#
# So, some complete examples are:
# e&m=1-12
# nethdlc=13-24
# fxsls=25,26,27,28
# fxols=29-32
#
#fxoks=1-24
#bchan=25-47
#dchan=48
#fxols=1-12
#fxols=13-24
#e&m=25-29
#nethdlc=30-33
#clear=44
#clear=45
#clear=46
#clear=47
#fcshdlc=48
#dacs=1-24:48
#dacsrbs=1-24:48
#
# Finally, you can preload some tone zones, to prevent them from getting
# overwritten by other users (if you allow non-root users to open /dev/zap/*
# interfaces anyway. Also this means they won't have to be loaded at runtime.
# The format is "loadzone=<zone>" where the zone is a two letter country code.
#
# You may also specify a default zone with "defaultzone=<zone>" where zone
# is a two letter country code.
#
# An up-to-date list of the zones can be found in the file zaptel/zonedata.c
#
loadzone = us
#loadzone = us-old
#loadzone=gr
#loadzone=it
#loadzone=fr
#loadzone=de
#loadzone=uk
#loadzone=fi
#loadzone=jp
#loadzone=sp
#loadzone=no
#loadzone=hu
#loadzone=lt
#loadzone=pl
defaultzone=us
#
# Section for PCI Radio Interface
# (see http://www.zapatatelephony.org/app_rpt.html)
#
# The PCI Radio Interface card interfaces up to 4 two-way radios (either
# a base/mobile radio or repeater system) to Zaptel channels. The driver
# may work either independent of an application, or with it, through
# the driver;s ioctl() interface. This file gives you access to specify
# load-time parameters for Radio channels, so that the driver may run
# by itself, and just act like a generic Zaptel radio interface.
#
# Unlike the rest of this file, you specify a block of parameters, and
# then the channel(s) to which they apply. CTCSS is specified as a frequency
# in tenths of hertz, for example 131.8 HZ is specified as 1318. DCS
# for receive is specified as the code directly, for example 223. DCS for
# transmit is specified as D and then the code, for example D223.
#
# The hardware supports a "community" CTCSS decoder system that has
# arbitrary transmit CTCSS or DCS codes associated with them, unlike
# traditional "community" systems that encode the same tone they decode.
#
# this example is a single tone DCS transmit and receive
#
# # specify the transmit tone (in DCS mode this stays constant)
# tx=D371
# # specify the receive DCS code
# dcsrx=223
#
# this example is a "community" CTCSS (if you only want a single tone, then
# only specify 1 in the ctcss list)
#
# # specify the default transmit tone (when not receiving)
# tx=1000
# # Specify the receive freq, the tag (use 0 if none), and the transmit code.
# # The tag may be used by applications to determine classification of tones.
# # The tones are to be specified in order of presedence, most important first.
# # Currently, 15 tones may be specified..
# ctcss=1318,1,1318
# ctcss=1862,1,1862
#
# The following parameters may be omitted if their default value is acceptible
#
# # set the receive debounce time in milliseconds
# debouncetime=123
# # set the transmit quiet dropoff burst time in milliseconds
# bursttime=234
# # set the COR level threshold (specified in tenths of millivolts)
# # valid values are {3125,6250,9375,12500,15625,18750,21875,25000}
# corthresh=12500
# # Invert COR signal {y,n}
# invertcor=y
# # set the external tone mode; yes, no, internal {y,n,i}
# exttone=y
#
# Now apply the configuration to the specified channels:
#
# # We are all done with our channel parameters, so now we specify what
# # channels they apply to
# channels=1-4
fxoks=1
fxsks=4
-------------- next part --------------
;
; Zapata telephony interface
;
; Configuration file
;
; You need to restart Asterisk to re-configure the Zap channel
; CLI> reload chan_zap.so
; will reload the configuration file,
; but not all configuration options are
; re-configured during a reload.
[trunkgroups]
;
; Trunk groups are used for NFAS or GR-303 connections.
;
; Group: Defines a trunk group.
; group => <trunkgroup>,<dchannel>[,<backup1>...]
;
; trunkgroup is the numerical trunk group to create
; dchannel is the zap 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 => <zapspan>,<trunkgroup>[,<logicalspan>]
;
; zapspan is the zap 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
;
; Default context
;
context=default
;
; Switchtype: Only used for PRI.
;
; national: National ISDN 2 (default)
; dms100: Nortel DMS100
; 4ess: AT&T 4ESS
; 5ess: Lucent 5ESS
; euroisdn: EuroISDN
; ni1: Old National ISDN 1
; qsig: Q.SIG
;
;switchtype=national
;
; Some switches (AT&T especially) require network specific facility IE
; supported values are currently 'none', 'sdn', 'megacom', 'accunet'
;
;nsf=none
;
; PRI Dialplan: Only RARELY used for PRI.
;
; unknown: Unknown
; private: Private ISDN
; local: Local ISDN
; national: National ISDN
; international: International ISDN
;
;pridialplan=national
;
; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's numbering plan)
;
; unknown: Unknown
; private: Private ISDN
; local: Local ISDN
; national: National ISDN
; international: International ISDN
;
;prilocaldialplan=national
;
; PRI callerid prefixes based on the given TON/NPI (dialplan)
; This is especially needed for euroisdn E1-PRIs
;
; 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 channels, defaults to 3600
; minimum 60 seconds
; some PBXs don't like channel restarts. so set the interval to a very long interval e.g. 100000000
; or 'never' to disable *entirely*.
;
;resetinterval = 3600
;
; Overlap dialing mode (sending overlap digits)
;
;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
;
; 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 = 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)
;
; 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.
; facilityenable = yes
;
;
; Signalling method (default is fxs). Valid values:
; em: E & M
; em_w: E & M Wink
; featd: Feature Group D (The fake, Adtran style, DTMF)
; featdmf: Feature Group D (The real thing, MF (domestic, US))
; featdmf_ta : Feature Group D (The real thing, MF (domestic, US)) through a Tandem Access point
; featb: Feature Group B (MF (domestic, US))
; fxs_ls: FXS (Loop Start)
; fxs_gs: FXS (Ground Start)
; fxs_ks: FXS (Kewl Start)
; fxo_ls: FXO (Loop Start)
; fxo_gs: FXO (Ground Start)
; fxo_ks: FXO (Kewl Start)
; pri_cpe: PRI signalling, CPE side
; pri_net: PRI signalling, Network side
; 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
; 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)
;
;signalling=fxo_ls
;
; For Feature Group D Tandem access, to set the default CIC and OZZ use
; these parameters:
;defaultozz=0000
;defaultcic=303
;
; A variety of timing parameters can be specified as well
; Including:
; 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)
;
rxwink=300 ; Atlas seems to use long (250ms) winks
;
; How long generated tones (DTMF and MF) will be played on the channel (in miliseconds)
;toneduration=100
;
; Whether or not to do distinctive ring detection on FXO lines
;
;usedistinctiveringdetection=yes
;
; Whether or not to use caller ID
;
usecallerid=yes
;
; Type of caller ID signalling in use
; bell = bell202 as used in US, v23 = v23 as used in the UK, dtmf = DTMF as used in Denmark, Sweden and Netherlands
;
;cidsignalling=bell
;
; What signals the start of caller ID
; ring = a ring signals the start, polarity = polarity reversal signals the start
;
;cidstart=ring
;
; Whether or not to hide outgoing caller ID (Override with *67 or *82)
;
hidecallerid=no
;
; Whether or not to enable call waiting on FXO lines
;
;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
;
usecallingpres=yes
;
; Some countries (UK) have ring tones with different ring tones (ring-ring),
; which means the callerid needs to be set later on, and not just after
; the first ring, as per the default.
;
;sendcalleridafter=1
;
;
; Support Caller*ID on Call Waiting
;
;callwaitingcallerid=yes
;
; Support three-way calling
;
;threewaycalling=yes
;
; Support flash-hook call transfer (requires three way calling)
; Also enables call parking (overrides the 'canpark' parameter)
;
;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)
;
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 recieved 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 at 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.
;
echocancel=yes
;
; Generally, it is not necessary (and in fact undesirable) to echo cancel
; when the circuit path is entirely TDM. You may, however, reverse this
; behavior by enabling the echo cancel during pure TDM bridging below.
;
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
; asterisk to briefly mute the channel, send an impulse, and use the impulse
; response to pre-train the echo canceller so it can start out with a much
; closer idea of the actual echo. Value may be "yes", "no", or a number of
; milliseconds to delay before training (default = 400)
;
;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)
;
rxgain=0.0
txgain=0.0
;
; Logical groups can be assigned to allow outgoing rollover. Groups
; range from 0 to 63, and multiple groups can be specified.
;
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
;
callgroup=1
pickupgroup=1
;
; Specify whether the channel should be answered immediately or
; if the simple switch should provide dialtone, read digits, etc.
;
immediate=no
;
; Specify whether flash-hook transfers to 'busy' channels should complete
; or return to the caller performing the transfer (default is yes).
;
;transfertobusy=no
;
; CallerID can be set to "asreceived" or a specific number
; if you want to override it. Note that "asreceived" only
; applies to trunk interfaces.
;
;callerid=2564286000
;
; 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
;
; 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, is also possible to specify how many
; busy tones to wait for before hanging up. The default is 4, but
; better results can be achieved if set to 6 or even 8. Mind that
; higher the number, more time is needed to hangup a channel, but
; lower is probability to get random hangups
;
;busycount=4
;
; If busydetect is enabled, is also possible to specify the
; cadence of your busy signal. In many countries it is 500mec
; on, 500msec off.
; Without busypattern specified, we'll accept any regular
; sound-silence pattern than 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 the Asterisk Makefile you'll find further options to tweak
; the busy detector. If your country has a busy tone with the same
; lengh tone and silence (as many countries do), consider defining
; the -DBUSYDETECT_COMPARE_TONE_AND_SILENCE option.
;
; 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
;
; 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"
;
; 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=us
;
; FXO (FXS signalled) devices must have a timeout to determine whe there was a
; hangup before the line was answered. This value can be tweaked to shorten
; how long it takes before Zap considers a non-ringing line to have hungup.
;
;ringtimeout=8000
;
; For FXO (FXS signalled) devices, whether to use pulse dial instead of DTMF
;
;pulsedial=yes
;
; For fax detection, uncomment one of the following lines. The default is *OFF*
;
;faxdetect=both
;faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
;
; Select which class of music to use for music on hold. If not specified
; then the default will be used.
;
;musiconhold=default
;
; PRI channels can have an idle extension and a minunused number. So long
; as at least "minunused" channels are idle, chan_zap will try to call
; "idledial" on them, and then dump them into the PBX in the "idleext"
; extension (which is of the form exten at 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.
;
;idledial=6999
;idleext=6999 at dialout
;minunused=2
;minidle=1
;
; Configure jitter buffers in zapata (each one is 20ms, default is 4)
;
;jitterbuffers=4
;
; 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 callerid
; spill is to be placed. Also, if you define any custom cadences, the
; default cadences will be turned off.
;
; 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" <(256) 428-6123>
;callerid="CallerID Phone" <(630) 372-1564>
;callerid="CallerID Phone" <(256) 704-4666>
;channel => 3
;callerid="Pac Tel Phone" <(256) 428-6124>
;channel => 4
;callerid="Uniden Dead" <(256) 428-6125>
;channel => 5
;callerid="Cortelco 2500" <(256) 428-6126>
;channel => 6
;callerid="Main TA 750" <(256) 428-6127>
;channel => 44
;
; For example, maybe we have some other channels
; which start out in a different context and use
; E & M signalling instead.
;
;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 = national
; signalling = pri_cpe
; group = 2
; channel => 1-23
;
; Used for distintive ring support for x100p.
; You can see the dringX patterns is to set any one of the dringXcontext fields
; and they will be printed on the console when an inbound call comes in.
;
;dring1=95,0,0
;dring1context=internal1
;dring2=325,95,0
;dring2context=internal2
; If no pattern is matched here is where we go.
;context=default
;channel => 1
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
context=default
callerid="Analog Phone"
signalling=fxo_ks
channel=>1
context=incoming
signalling=fxs_ks
callerid=asreceived
channel=>4
-------------- next part --------------
;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
; This configuration file is reloaded
; - With the "extensions reload" command in the CLI
; - With the "reload" command (that reloads everything) in the CLI
;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no
;
; If autofallthrough is set, then if an extension runs out of
; things to do, it will terminate the call with BUSY, CONGESTION
; or HANGUP depending on Asterisk's best guess (strongly recommended).
;
; If autofallthrough is not set, then if an extension runs out of
; things to do, asterisk will wait for a new extension to be dialed
; (this is the original behavior of Asterisk 1.0 and earlier).
;
autofallthrough=yes
;
; If clearglobalvars is set, global variables will be cleared
; and reparsed on an extensions reload, or Asterisk reload.
;
; If clearglobalvars is not set, then global variables will persist
; through reloads, and even if deleted from the extensions.conf or
; one if its included files, will remain set to the previous value.
;
clearglobalvars=no
;
; If priorityjumping is set to 'yes', then applications that support
; 'jumping' to a different priority based on the result of their operations
; will do so (this is backwards compatible behavior with pre-1.2 releases
; of Asterisk). Individual applications can also be requested to do this
; by passing a 'j' option in their arguments.
;
priorityjumping=no
;
; You can include other config files, use the #include command (without the ';')
; Note that this is different from the "include" command that includes contexts within
; other contexts. The #include command works in all asterisk configuration files.
;#include "filename.conf"
; The "Globals" category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
OUTBOUNDTRUNK=Zap/4
[default]
include => incoming
include =>internal
[incoming]
exten => s,1,Wait(1)
exten => s,2,Dial(Zap/1|20,t)
exten => s,3,Voicemail(u1000 at default)
exten => s,4,Hangup
exten => s,103,Voicemail(b1000 at default)
exten => s,104,Hangup
[internal]
include => outbound-local
include => outbound-long-distance
exten => 1000,1,Dial,Zap/1|10
exten => 1000,2,Voicemail(u1000 at default)
exten => 1000,3,Hangup
exten => 1000,102,Voicemail(b1000 at default)
exten => 1000,103,Hangup
exten => 1001,1,Dial(SIP/colin2,10)
exten => 1001,2,Voicemail(u1001 at default)
exten => 1001,3,Hangup
exten => 1001,102,Voicemail(b1001 at default)
exten => 1001,103,Hangup
exten => 1212,1,Dial(SIP/colin,10)
exten => 1212,2,Voicemail(u1212 at default)
exten => 1212,3,Hangup
exten => 1212,102,Voicemail(b1212 at default)
exten => 1212,103,Hangup
exten => 1811,1,Dial(SIP/peder,10)
exten => 1811,2,Voicemail(u1811 at default)
exten => 1811,3,Hangup
exten => 1811,102,Voicemailb1811 at default)
exten => 1811,103,Hangup
exten => 1789,1,Dial(SIP/brad,10)
exten => 1789,2,Voicemail(u1789 at default)
exten => 1789,3,Hangup
exten => 1789,102,Voicemail(b1789 at default)
exten => 1789,103,Hangup
;extension to dial to access VM
exten => 5111,1,VoiceMailMain()
[outbound-local]
exten => _9NXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten => _9NXXXXXX,2,Congestion()
exten => _9NXXXXXX,102,Congestion()
exten => 911,1,Dial(${OUTBOUNDTRUNK}/911)
exten => 9911,1,Dial(${OUTBOUNDTRUNK}/911)
[outbound-long-distance]
exten => _91NXXNXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten => _91NXXNXXXXXX,2,Congestion()
exten => _91NXXNXXXXXX,102,Congestion()
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