Archive > Asterisk

I can make an outside call, but can't recieve

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archived:
First of all I would like to thank the developers for this wonderful piece of work.

I have installed pluto on a machine with Digum card with one fxo module. I managed to make an outside call by changing the configuration in Zaptel.conf and zapata.conf files. However, I failed to make the system recieve calls. Any suggestion how to solve this problem!

The other thing I would like to do is to install webmin on the system for more control. Can someone help and direct me on how to do that?

Thanks in advance.

archived:
it should work to receive calls also.


--- Quote ---The other thing I would like to do is to install webmin on the system for more control. Can someone help and direct me on how to do that?
--- End quote ---

what exactly are trying to do? i don't understand.

archived:
Thak you very much for your reoly.

I am new to linux and learning my way through. I know "webmin" ( newbielink:http://www.webmin.com [nonactive]) is a great grapgical tool to control linux servers and systems. I have installed and used this software on Centos when I was using asterisk from the tixbox dist. it gave better interface for control rather than typing thousands of command lines. I wish I can install webmin on plutohome system too.

For recieving calls, I have tried changing many parameters in zaptel.conf and zapata.conf files to make it work, but without success. Do I need to change other zonf files other than these two? please find below my two files :

zaptel.conf

#
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
#
# First come the span definitions, in the format
# span=<span num>,<timing source>,<line build out (LBO)>,<framing>,<coding>[,yellow]
#
# All T1/E1 spans generate a clock signal on their transmit side. The
# <timing source> parameter determines whether the clock signal from the far
# end of the T1/E1 is used as the master source of clock timing. If it is, our
# own clock will synchronise to it. T1/E1's connected directly or indirectly to
# a PSTN provider (telco) should generally be the first choice to sync to. The
# PSTN will never be a slave to you. You must be a slave to it.
#
# Choose 1 to make the equipment at the far end of the E1/T1 link the preferred
# source of the master clock. Choose 2 to make it the second choice for the master
# clock, if the first choice port fails (the far end dies, a cable breaks, or
# whatever). Choose 3 to make a port the third choice, and so on. If you have, say,
# 2 ports connected to the PSTN, mark those as 1 and 2. The number used for each
# port should be different.
#
# If you choose 0, the port will never be used as a source of timing. This is
# appropriate when you know the far end should always be a slave to you. If the
# port is connected to a channel bank, for example, you should always be its
# master. Any number of ports can be marked as 0.
#
# Incorrect timing sync may cause clicks/noise in the audio, poor quality or failed
# faxes, unreliable modem operation, and is a general all round bad thing.
#
# 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
#
fxsks=4
#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 newbielink:http://www.zapatatelephony.org/app_rpt.html [nonactive])
#
# 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


zapata.conf

;
; 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
; passthrough:     Listen to the telco
;
; priindication = outofband
;
; PRI/BRI transfers (HOLD -> SETUP -> ECT/Hangup)
;
; Configure how transfers are initiated. ECT should be preferred
;
; no:      no transfers allowed (results in hangup)
; ect:     use ECT (facility)
: hangup:   transfer on hangup (if your phones dont support ECT)
;
; pritransfer = ect
;
; 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=fxs_ks
;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@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=0
;
; Ring groups (a.k.a. call groups) and pickup groups.  If a phone is ringing
; and it is a member of a group which is one of your pickup groups, then
; you can answer it by picking up and dialing *8#.  For simple offices, just
; make these both the same.  Groups range from 0 to 63.
;
callgroup=0
pickupgroup=0

;
; 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, it 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 the higher the number, the more
; time that will be needed to hangup a channel, but lowers the probability
; that you will get random hangups.
;
;busycount=4
;
; If busydetect is enabled, it is also possible to specify the cadence of your
; busy signal.  In many countries, it is 500msec on, 500msec off.  Without
; busypattern specified, we'll accept any regular sound-silence pattern that
; repeats <busycount> times as a busy signal.  If you specify busypattern,
; then we'll further check the length of the sound (tone) and silence, which
; will further reduce the chance of a false positive.
;
;busypattern=500,500
;
; NOTE: In the Asterisk Makefile you'll find further options to tweak the busy
; detector.  If your country has a busy tone with the same length tone and
; silence (as many countries do), consider 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 if 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@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@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
channel => 4



the other problem i am facing now is that everytime I reboot the system, asterisk stops working. I have to follow the following procedure to make work again:

1) ztcfg -vvvv
2) /etc/init.d/asterisk restart
3) asterisk -r
4) reload chan_zap.so
5) reload
6) exit

Thank you very much for your support.

archived:
To recieve incomming calls you need to change the context=default to context=from-pstn in zapata.conf

NOS.

archived:
YES it worked :)

Thank you very very much.

Any help regarding webmin?

Do you have any idea about the message ( Got SIP response 841 "Subscription Does Not Exist" back from xxx.xxx.xxx.xxx ), and how to get red of it?

The other question, why asterisk stops working when I restart the system. I have to restart it manually and after I issue the command ztcfg -vvvv. Is there any way to solve this problem?

Thank you in advance.

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