ooH323 no calls from h323 to sip

Hi,
I set up asterisk 1.6 on my debian linux machine. I installed asterisk-addons and GNUgk 2.2.7. I configured a SIP softphone (200), and two H323 hardware phones (101,102). I can call SIP=>H323, H323=>SIP doesn’t work. Can someone tell me what I configured wrong? Here’s my config files…

gatekeeper.ini

[Gatekeeper::Main]
Fortytwo=42
Name=GK
Home=192.168.1.127
Bind=192.168.1.127
TimeToLive=300
CompareAliasType=0
TraceLevel=2
StatusPort=7000
TimestampFormat=RFC822
EncryptAllPasswords=0
UseBroadcastListener=1
UnicastRasPort=1719
UseMulticastListener=1
MulticastPort=1718
MulticastGroup=224.0.0.1
EndpointSignalPort=1720
ListenQueueLength=1024

[RoutedMode]
GkRouted=1
H245Routed=1
CallSignalPort=1721
AcceptUnregistredCalls=1

[RoutingPolicy]

extensions.conf
[default]
exten => 1001,1,Answer()
exten => 1001,2,Playback(hello-world)
extem => 1001,3,Hangup()

exten => 102,1,Dial(OOH323/102@192.168.1.24)
exten => 101,1,Dial(OOH323/101@192.168.1.22)
exten => 200,1,Dial(sip/200)

ooh323.conf
[general]
bindaddr=192.168.1.127
h323id=asteriskH323
e164=100

gateway=yes
gatekeeper=192.168.1.127

context=default
accountcode=default
disallow=all

allow=g729
allow=g731
allow=gsm
allow=ulaw

dtmfmode=q931keypad

please excuse stupid mistakes - I’m new to asterisk (it took me some time to get this far, but now I’m stuck)