[asterisk-users] RTP Stream not set up correct at outgoing call

[ontae]

Dear "astreisk-users mailing list" subscribers,

with my asterisk under debian (Version: 1:1.0.9.dfsg-5) i have the problem, that
the RTP stream is not set up correct with an outgoing call. Incoming calls are
working with no problems.

The problem is, that the RTP stream is initiated from IP A and my asterisk or my
router/with firewall (ISA 2004) replies to the port the RTP stream comes from,
but to the ip-adress it is talking SIP, not the IP A, from where the RTP stream
is set up.
For a better understaning please see
"http://www.ontae.net/tmp/graph_outgoing-call.JPG" or
"http://www.ontae.net/tmp/graph_outgoing-call.txt".
More infos can be found in the trace i took at my router/firewall at the
external interface (myIP):
"http://www.ontae.net/tmp/outgoing-call.trace.sip.txt".

Please have a look at my problem and give me a feedback.

Thanks in advance,
ontae