[Asterisk-Users] Analog calls not working

[Clive Carter]

Hi
I have battled my way through setting up linux,
and then installing Asterisk. I have got 90% of the way there.
Asterisk registers with my IAX provider ok, my SIP phones can send and 
receive calls
to each other, and out over the network.
Voicemail is working ok.
The last thing I want is to direct incoming analog calls to a SIP phone,
and to send all calls starting with a '9' out through my analog line.
I have been scratching my head over this for 2 days, and cannot find the 
answer anywhere.
Can anyone help please ?
 
This is the message I get when I try -
----------------------------------------------------------------------------
CLI output

Nov 10 15:45:13 DEBUG[81926]: Check for res for 2001
Nov 10 15:45:13 DEBUG[81926]: Call from user '2001' is 1 out of 0
Nov 10 15:45:13 DEBUG[81926]: build_route: Contact hop: 
<sip:2001 at 192.168.1.211;user=phone>
Nov 10 15:45:13 VERBOSE[294931]:
[1;37;40mAsterisk Ready.[0;37;40m    -- Executing 
[1;36;40mDial[0;37;40m("[1;35;40mSIP/2001-3d7a[0;37;40m", 
"[1;35;40mZap/1/07970856261[0;37;40m") in new stack
Nov 10 15:45:13 NOTICE[294931]: Unable to create channel of type 'Zap'
Nov 10 15:45:13 VERBOSE[294931]:   == Everyone is busy at this time

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

result of zap show channels
Chan     Extension    Context        Language    MusicOnHold
1            inbound-analog    en

---------------------------------------------------------------------
result of zap show channel     1
File Descriptor :         28
Span:            1I
Extension:    
Caller ID string:        no
Destroy:            0
Signalling Type:        FXS Kewlstart
Owner:            <None>
Real:            <None>
Callwait:            <None>
Threeway:        <None>
Confno:            -1
Propagated Conference:    -1
Real in Conference:        0
DSP:            noI>
Relax DTMF:        yes
Dialing/CallwaitCAS        0/0
Default law:        ulaw
Fax Handled:        no
Pulse Phone:        no
Echo Cancellation:        128 taps, currently off
Actual Confinfo:        Num/0, Mode/0x0000
Actual Confmute:        No
-------------------------------------------------
ZTCFG result
Channel Map:
Channel 01: FXS Kewlstart  (Default)  (Slaves: 01)
1 channels configured
---------------------------------------------------------------

ZAPTEL.CONF
loadzone = uk
defaultzone=uk
fxsks=1

ZAPATA.CONF
signalling=fxs_ks
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
context=inbound-analog
channel => 1

SIP.CONF
[general]
port = 5060
bindaddr = 0.0.0.0
disallow=all
allow=ulaw
allow=alaw
allow=gsm
register => ID:PASSWD at SIPPROVIDER
context = internal

(rest is configuration for SIP Phones)


EXTENSIONS.CONF
.
.
[internal]     ; context for SIP phones
exten => 01952XXXXXX,1,Dial(Zap/1/${EXTEN})
exten => 01952XXXXXX,2,Hangup
.
.
[inbound-analog]
exten => _0[1-9].,1,Dial(${OFFICE},15,Ttm)  ;OFFICE is SIP phone
exten => _0[1-9].,2,VoiceMail(u${OFFICEVM})    ; OFFICEVM is mailbox for 
OFFICE phone
exten => _0[1-9].,3,Hangup

TIA
Clive