[asterisk-users] callback and bridge problem

[Adam KOSA]

Greetings,

i've been posted a message to this list in july, which had one response. 
  Thanks for that idea!  Unfortunately asterisk is only a hobby, and did 
not have much time dealing with the problem since.  My original letter 
was long, i wouldn't post it again, the archive url is

http://archives.free.net.ph/message/20070710.053008.c02209c0.en.html

Since than i've upgraded to 1.4.8 from 1.2 series, i thought this might 
help.  It did not.

Answering to the question asked from me in july, no, i'm not behind nat, 
  and i did not have reinvite=yes in my configs.  I've put it into the 
sip.conf, tried, but the call hung up again.

I'd be greatful for more ideas of solving the problem.

Fresh logs when hanging up, from asterisk console:

     -- SIP/neophonex99-out-08213ac8 is making progress passing it to 
SIP/neophonex57-out-081e8a78
[Aug  2 21:54:51] WARNING[24739]: chan_sip.c:11948 
handle_response_invite: Re-invite to non-existing call leg on other UA. 
SIP dialog '29312b1d2fc464560ff9ef7747c614d2 at sip.neophonex.hu'. Giving up.
     -- SIP/neophonex99-out-08213ac8 answered SIP/neophonex57-out-081e8a78
     -- Native bridging SIP/neophonex57-out-081e8a78 and 
SIP/neophonex99-out-08213ac8
[Aug  2 21:54:57] WARNING[24739]: chan_sip.c:11948 
handle_response_invite: Re-invite to non-existing call leg on other UA. 
SIP dialog '6d52c03e65d0dfaa53434ea81b948df6 at sip.neophonex.hu'. Giving up.
   == Spawn extension (internal, 9520620*********, 3) exited non-zero on 
'SIP/neophonex57-out-081e8a78'
[Aug  2 21:54:57] NOTICE[24749]: pbx_spool.c:351 attempt_thread: Call 
completed to SIP/0630********@neophonex57-out


Thanks for any help
Adam