asterisk users - Oct 2005 - dial-out gives always "not found" (dial-in works fine)

Hi,

I'm running asterisk 1.0.9 with an FXO card. People can call me on my pstn
line and that gets transferred to my laptop (on 192.168.62.100). That all runs
fine.
If, though, I want to dial out (a pstn line) I always get a "call rejected:
404 not found" error (in sjphone) or a plain "not found" (in
kphone) and in the asterisk console this appears:
Check for res for
is not a local user
is not a local user
Stopping retransmission on
'66ED2B84-1DD2-11B2-8B72-D132D0F7B1FD@192.168.62.100' of Response 1:
Found

In sip.conf I have this:
[1000]
type=peer
host=dynamic
defaultip=192.168.62.100
dtmfmode=rfc2833
mailbox=0000
context=dialoutcont
callerid="Folkert van Heusden" <folkert@keetweej.vanheusden.com>

and in extensions.conf:
[dialoutcont]
exten => _0XXXXXXXXX,1,Dial(ZAP/1/${EXTEN})

Anyone got a suggestion what might be wrong?


Folkert van Heusden

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On Mon, 31 Oct 2005, Folkert van Heusden wrote:
> In sip.conf I have this:
> [1000]
> type=peer
> host=dynamic
> defaultip=192.168.62.100
> dtmfmode=rfc2833
> mailbox=0000
> context=dialoutcont
> callerid="Folkert van Heusden"
<folkert@keetweej.vanheusden.com>
You've defined a peer - which is for calls TO that endpoint.  You need a 
"user" if you want to accept calls FROM that endpoint.  Or, use a
"friend"
if you want both and all the settings can be the same.

Steve