Author Topic: I can make an outside call, but can't recieve  (Read 18371 times)


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I can make an outside call, but can't recieve
« on: November 16, 2006, 11:51:17 pm »
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.


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I can make an outside call, but can't recieve
« Reply #1 on: November 17, 2006, 09:38:32 am »
it should work to receive calls also.

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?

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


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I can make an outside call, but can't recieve
« Reply #2 on: November 17, 2006, 08:31:31 pm »
Thak you very much for your reoly.

I am new to linux and learning my way through. I know "webmin" ( 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 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.
# 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
# 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
# Section for PCI Radio Interface
# (see
# 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 telephony interface
; Configuration file
; You need to restart Asterisk to re-configure the Zap channel
; CLI> reload
;      will reload the configuration file,
;      but not all configuration options are
;       re-configured during a reload.

; 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

; Default language
; Default context
; 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
; Some switches (AT&T especially) require network specific facility IE
; supported values are currently 'none', 'sdn', 'megacom', 'accunet'
; PRI Dialplan:  Only RARELY used for PRI.
; unknown:        Unknown
; private:        Private ISDN
; local:          Local ISDN
; national:     National ISDN
; international:  International ISDN
; 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
; 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)
; 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)
; For Feature Group D Tandem access, to set the default CIC and OZZ use these
; parameters:
; 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)
; Whether or not to do distinctive ring detection on FXO lines

; Whether or not to use caller ID
; 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
; What signals the start of caller ID
;     ring     = a ring signals the start
;     polarity = polarity reversal signals the start
; Whether or not to hide outgoing caller ID (Override with *67 or *82)
; Whether or not to enable call waiting on FXO lines
; Whether or not restrict outgoing caller ID (will be sent as ANI only, not
; available for the user)
; Mostly use with FXS ports
; Whether or not use the caller ID presentation for the outgoing call that the
; calling switch is sending.
; 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.
; Support Caller*ID on Call Waiting
; Support three-way calling
; Support flash-hook call transfer (requires three way calling)
; Also enables call parking (overrides the 'canpark' parameter)
; Allow call parking
; ('canpark=no' is overridden by 'transfer=yes')
; Support call forward variable
; Whether or not to support Call Return (*69)
; 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:
; for any other voicemail context, the following will produce the stutter tone:
; 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.
; 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.
; 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)
; 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.
; You may also set the default receive and transmit gains (in dB)
; Logical groups can be assigned to allow outgoing rollover.  Groups range
; from 0 to 63, and multiple groups can be specified.
; 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.

; Specify whether the channel should be answered immediately or if the simple
; switch should provide dialtone, read digits, etc.
; Specify whether flash-hook transfers to 'busy' channels should complete or
; return to the caller performing the transfer (default is yes).
; CallerID can be set to "asreceived" or a specific number if you want to
; override it.  Note that "asreceived" only applies to trunk interfaces.
; AMA flags affects the recording of Call Detail Records.  If specified
; it may be 'default', 'omit', 'billing', or 'documentation'.
; Channels may be associated with an account code to ease
; billing
; ADSI (Analog Display Services Interface) can be enabled on a per-channel
; basis if you have (or may have) ADSI compatible CPE equipment
; 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.
; 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.
; 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.
; 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
; Use a polarity reversal to mark when a outgoing call is answered by the
; remote party.
; 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.
; 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.
; 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.
; For FXO (FXS signalled) devices, whether to use pulse dial instead of DTMF
; For fax detection, uncomment one of the following lines.  The default is *OFF*
; Select which class of music to use for music on hold.  If not specified
; then the default will be used.
; 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.
; Configure jitter buffers in zapata (each one is 20ms, default is 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:
; 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.
;channel => 15
;channel => 16

; All those in group 0 I'll use for outgoing calls
; Strip most significant digit (9) before sending
;channel => 45

;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.
; If no pattern is matched here is where we go.
;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
5) reload
6) exit

Thank you very much for your support.


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I can make an outside call, but can't recieve
« Reply #3 on: November 18, 2006, 06:57:26 pm »
To recieve incomming calls you need to change the context=default to context=from-pstn in zapata.conf



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I can make an outside call, but can't recieve
« Reply #4 on: November 26, 2006, 11:26:26 pm »
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 ), 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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I can make an outside call, but can't recieve
« Reply #5 on: November 27, 2006, 03:00:35 pm »

I have the same problem with the zap card not being initialised correctly at startup.  I will have a look and see if I can find a solution.

Sorry cant help with webmin as I have not tried to install it on the pluto distribution.



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Install Webmin on PLUTOHOME
« Reply #6 on: January 24, 2007, 07:08:27 pm »
Hi all,

I have just installed WEBMIN now on my fresh PlutoHome core:

here is how I have done:

From my WinXP (sorry... sorry... sorry... sorry... )

1 - download the latest file from

2 - upload the file to my core (I put it on /home/userxxx/download/)

3 - login to the core as root (again sorry....)

4 - go to the folder where you have the downloaded file

5 - run the command "dpkg --install webmin_1.320_all.deb"
wait for it till it ends... with errors

6 - run the command apt-get -f install

7 - connect to webmain http://YourMachineIp:10000

8 - enjoy it


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Great minds think alike :)
« Reply #7 on: February 01, 2007, 02:39:22 pm »
Thank you very much for reply. I have just figured out how to that two days ago. I did exactly what you did. I was about to post my findings, but you were faster :)

Thanks again


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I can make an outside call, but can't recieve
« Reply #8 on: February 08, 2007, 11:07:10 am »
could you please write a wiki page explaining the steps? maybe other users will be interested in this.
thank you