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
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