[asterisk-users] How to test my Dial(SIP/...) ?

[tjoen]

Or how to get my Dial(SIP/...) working? I am new in Asterisk.
All other Asterisk tests worked, exception is Dial(SIP/..

My setup:
ADSL NAT Router has UDP ports 5060 to 5070 and 8766 to 35000
forwarded to 192.168.254.1

On 192.168.254.1. Linux system from sources. Asterisk 1.6.2.1
I know that there are codecs missing.
Adjusted sip.conf accordingly.
Registering myself (with "register =>" in sip.conf) to ekiga.net
succeeded too.
sip.conf:
[general]
context=default
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
sipdebug = yes
qualify=yes
externip=195.241.23.211
localnet=192.168.254.0/255.255.255.0
nat=yes
[basic-options](!)
        dtmfmode=rfc2833
        context=from-office
        type=friend
[natted-phone](!,basic-options)
        nat=yes
        directmedia=no
        host=dynamic
[2131](natted-phone)
        secret = ******
        disallow=all 
	allow=g729
        allow=gsm
        allow=g723
        allow=ulaw

extensions.conf:
[general]
static=yes
writeprotect=no
clearglobalvars=no
[globals]
; all until and including [default] from make samples
[default]
include => demo
[from-office]
exten => 1020,1,Dial(SIP/500 at ekiga.net)
include => default

On 192.168.254.2 Linux from sources with Ekiga 3.2.6
succeeds registering as [2131]. Demo, IAX, echo and
console tests passed. But never got SIP working from Ekiga.
No errors in /var/log/messages

On www.tjoen.dds.nl/asterisk.log the output of
# asterisk -vvv 
with only one attempt to dial 1020

Looks like the "line" is busy? 
Or an error in my setup?
Do I need something more to get SIP working?