asterisk users - Mar 2004 - SIP - Native Bridging - sipgate.de

Hi,

I have the following two tries. While the first calls out just fine, the 
second call does not work.

My asterisk box is behind a NATed fli4l fw. Port 10000-20000, 5060, 5004 
are all forwarded to the asterisk box.

What throws me off, is the fact, that the first call worked fine, and 
the second did not.

extension.conf
=============exten => _*9.,1,Dial,SIP/${EXTEN:2}@sipgate,30,tr

sip.conf
=======[general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0              ; Address to bind to
context = default               ; Default for incoming calls
nat = yes
canreinvite=no
register => 4318776:xxxxxx@sipgate.de/4318776

[sipgate]
secret=xxxxxxx
username=4318776
fromuser=4318776
fromdomain=sipgate.net
type=friend
host=sipgate.de
nat=yes
dtmfmode=rfc2833
canreinvite=no
context=in-sipgate


Any and all help is greatly apprciated.

     -- Executing Dial("SIP/25-ebe3",
"SIP/01729731418@sipgate") in new
stack
     -- Called 01729731418@sipgate
     -- SIP/sipgate-db58 is making progress passing it to SIP/25-ebe3
   == Spawn extension (default, *901729731418, 1) exited non-zero on 
'SIP/25-ebe3'
     -- Executing Dial("SIP/25-e2f9",
"SIP/01715828552@sipgate") in new
stack
     -- Called 01715828552@sipgate
     -- SIP/sipgate-e2ee is making progress passing it to SIP/25-e2f9
     -- SIP/sipgate-e2ee answered SIP/25-e2f9
     -- Attempting native bridge of SIP/25-e2f9 and SIP/sipgate-e2ee
   == Spawn extension (default, *901715828552, 1) exited non-zero on 
'SIP/25-e2f9

--
Best regards

Peer Oliver Schmidt
the internet company

I hate to answer myself, but another thing:
 > The first call worked fine.

This is not true. The moment the other side answers the call, Asterisk
     -- Attempting native bridge of SIP/25-e2f9 and SIP/sipgate-e2ee

This obviously can't work. How can I make sure asterisk never tries to 
do this?!

canreinvite is set to no

TIA
--
Best regards

Peer Oliver Schmidt
the internet company