[Asterisk-Users] problem with running ztcfg

Paul Chan wayofdragon at yahoo.com
Wed Feb 9 12:12:16 MST 2005


I am using X100P and the files that come with the
samples (asterisk 1.0.5): 

zaptel.conf:

-----------------------------------------------------
;
; Zapata telephony interface
;
; Configuration file
                                                      
                         
[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
;
; 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
;
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
;
; 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
;
;
; 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))
; 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))
; 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
;
; 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
;
; 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)
;
transfer=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 31, 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
;
; 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
;
;busydetect=yes
;
; If busydetect is enabled, is also possible to
specify how many
; busy tones to wait 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
;
; 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
;
; 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

-----------------------------------------------------

zapata.conf:

-----------------------------------------------------
loadzone = us
#loadzone=fr
#loadzone=de
#loadzone=uk
#loadzone=fi
#loadzone=jp
#loadzone=sp
#loadzone=no
defaultzone=us
#
fxsks=1

-----------------------------------------------------
--- Dana Olson <rickaster at gmail.com> wrote:

> On Wed, 9 Feb 2005 08:23:57 -0800 (PST), Paul Chan
> <wayofdragon at yahoo.com> wrote:
> > Hi All,
> > 
> >  I just installed Asterisk 1.0.5, and the
> > installation went fine (I ran modprobe zaptel and
> > modprobe wcfxo).  However, when I ran ztcfg I get
> the
> > following:
> > 
> > ioctl(ZT_LOADZONE) failed: Invalid argument
> > Notice: Configuration file is /etc/zaptel.conf
> > line 135: Unable to register tone zone 'us'
> > 
> >  After that I ran Asterisk and it seem to started
> ok,
> > except that it won't pick up any calls.  Has
> anyone
> > seen this or know what could be the problem?
> > 
> >  Thanks for your help in advance!
> 
> 
> What hardware are you using? What is your
> zaptel.conf and zapata.conf?
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
>
http://lists.digium.com/mailman/listinfo/asterisk-users
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>   
>
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> 


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