asterisk users - Jul 2006 - problem with iax -> sip across 2 asterisks

Hi,

I've got a bit of a puzzle with my setup. I've got 2 asterisks who are 
connected using dundi over iax. Both asterisks use the realtime stuff to 
fetch sip and iax client information from a mysql database. As long as all 
clients are sip, everything works great. With iax clients however I've got a
problem.
Situation is as follows: 
I've registered an iax softphone (I tried both idefisk and iaxComm, same 
result) with number 1003 on server hub1. I've registered a sip client with 
number 1001 on server hub2. 
If the iax client calls 1001, something goes wrong with native bridging.
If I pick up the call of the sip phone (1001), then I can see the asterisk 
where 1003 is registered (hub1) going for native bridging and thinks it 
works, but the iax softphone keeps ringing and has no clue the other side has 
picked up.
If the sip clients dials 1003, everything works just fine. This way hub1 also 
tries to native bridge the connections, but then everything works great.

This is what I see on the cli's in the situation where it goes wrong:
hub1 (has ip 10.10.1.31, iax client has ip 10.10.180.210):
    -- Accepting AUTHENTICATED call from 10.10.180.210:
       > requested format = gsm,
       > requested prefs = (),
       > actual format = gsm,
       > host prefs = (gsm|ulaw|alaw),
       > priority = mine
    -- Executing ChanIsAvail("IAX2/1003-1", "SIP/1001|sj")
in new stack
    -- Executing ChanIsAvail("IAX2/1003-1", "IAX2/1001|sj")
in new stack
    -- Executing Goto("IAX2/1003-1", "lookupdundi|1001|1")
in new stack
    -- Goto (lookupdundi,1001,1)
    -- Called priv:oIKeEUppyhc+NoU81ITgJQ@10.10.1.32/1001
    -- Call accepted by 10.10.1.32 (format gsm)
    -- Format for call is gsm
    -- IAX2/10.10.1.32:4569-3 is ringing
    -- IAX2/10.10.1.32:4569-3 stopped sounds
    -- IAX2/10.10.1.32:4569-3 answered IAX2/1003-1
    -- Attempting native bridge of IAX2/1003-1 and IAX2/10.10.1.32:4569-3
    -- Channel 'IAX2/10.10.1.32:4569-3' ready to transfer
    -- Channel 'IAX2/1003-1' ready to transfer
    -- Releasing IAX2/1003-1 and IAX2/10.10.1.32:4569-3
    -- Hungup 'IAX2/10.10.1.32:4569-3'
    -- Hungup 'IAX2/1003-1'

hub2 (has ip 10.10.1.32)
   -- Accepting AUTHENTICATED call from 10.10.1.31:
       > requested format = gsm,
       > requested prefs = (),
       > actual format = gsm,
       > host prefs = (),
       > priority = mine
    -- Executing Dial("IAX2/10.10.1.31:4569-1",
"SIP/1001||") in new stack
    -- Called 1001
    -- SIP/1001-081980d8 is ringing
    -- SIP/1001-081980d8 answered IAX2/10.10.1.31:4569-1
    -- Hungup 'IAX2/10.10.1.31:4569-1'

All servers and clients are in the same /16, no firewalls etc.
I tried Asterisk 1.2.7 and 1.2.10, where both have the same result.

I can fix it by disabling native bridging (canreinvite=no for the sip client 
and notransfer=yes for the iax client), but shouldn't Asterisk handle this 
situation without these settings.

Did I do anything wrong (or fail to understand something) or did I stumble 
across a bug?
Configs and/or output from "iax2 debug" are available on request!
Please cc me directly on any replies as I'm not (yet) subscribed to this
list!
Thanks in advance!!

Regards,

Guus Houtzager