[Asterisk-Users] dialout gives 404 (using sjphone (dialin works fine))

Folkert van Heusden folkert at vanheusden.com
Sun Oct 30 13:00:24 MST 2005


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) 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 at 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 at 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

-- 
Try MultiTail! Multiple windows with logfiles, filtered with regular
expressions, colored output, etc. etc. www.vanheusden.com/multitail/
----------------------------------------------------------------------
Get your PGP/GPG key signed at www.biglumber.com!
----------------------------------------------------------------------
Phone: +31-6-41278122, PGP-key: 1F28D8AE, www.vanheusden.com
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 282 bytes
Desc: Digital signature
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20051030/e8bf240d/attachment.pgp


More information about the asterisk-users mailing list