[Asterisk-Users] Why? oh why can't I dial out?

[Vassilis Konstantinou]

I have been struggling with my Asterisk setup for 3 days now and I think I 
have done well...apart from the small detail that I cannot dial out on my 
phone (PSTN) line.

My setup is:

Suse Linux 9.0
1 fxo card connected to a BT(UK) line
1 Cisco ATA186 sip v3.0 with two analogue phones attached to it
Asterix CVS-HEAD-05/30/04-06:56:31
with the UK Userid patch applied. Asterisk loads without any warnings.

The problem?
I can receive calls with the userid reported correctly. I can forward them 
to the two SIP ATA lines. I can dial internally (between the two phones) BUT
I cannot dial out :-(

I have tried everything (and yes I searched the world using google but 
nothing seems to apply to my case). So can somebody please direct me to 
possible causes.

The scenario is: if I dial 9123 (for the UK clock) then  output from the 
console is:

   -- Executing Dial("SIP/5000-96f1", "Zap/1/123") in new stack
     -- Called 1/123
     -- Zap/1-1 answered SIP/5000-96f1
     -- Hungup 'Zap/1-1'

SIP/5000 is one of my ATA phones
ZAP/1 is the fxo card

The call is transferred to Zap/1 as I can hear the dial tone but then 
nothing happens (it does not dial 123). It just stays on the tone until it 
times out. I also tried pressing buttons on my ATA phone but nothing is 
transferred. HELP!




Here is a collection of my conf files:

zaptel.conf

fxsks=1
loadzone=uk
defaultzone=uk

---------------------------
zapata.conf


[channels]
;
; Default language
;
language=en
;
; Default context
;
;context=default
context=incoming
switchtype=national
signalling=fxs_ks
usedistinctiveringdetection=no
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=no
transfer=no
ancallforward=no
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
relaxdtmf=yes
rxgain=0.0
txgain=4.0
immediate=no
musiconhold=default
busydetect=no
callprogress=no
usecallerid=uk

channel => 1

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

part of extensions.conf

[incoming]

exten => s,1,SetCallerId(${CALLERID})
exten => s,2,dial(SIP/5000&SIP/5001,10,tr)
exten => s,3,Voicemail,u1000
exten => s,102,Voicemail,b1000


exten => _9.,1,Dial(${CONSOLE}/${EXTEN:1})
exten => _9.,2,Congestion




Vassilis