Hi Users,
I'm here again.
After having read more docs I don't feel like using rtpproxy
My Asterisks have both public IP Adresses so I don't need to proxy rtp stream
I just need a configuration to randomly proxy SIP requests to
Asterisk boxes behind SER (for load balancing purposes)
and to handle registrations of user to forward incoming calls to them
(calls are coming from Asterisks to OpenSER)
I'm really stuck on it because I already have such a working setup,
with same config files.
I'm just trying to replicate it on another site
Tnx in advance for help
Edoardo
My last mail to the ml follow for reference
Tnx again Klaus,
i'll read docs more in depth and try to understand each
change to openser.cfg you suggested
U 2006/12/20
19:15:35.025446 OOO.OOO.OOO.OOO:5060 -> CCC.CCC.CCC.CCC:21722
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP
OOO.OOO.OOO.OOO:5060;branch=z9hG4bK-d87543-3e229802603d7c32-1--d87543-;rport=21722.
^^^^^^^^^^^^^^
strange bug again - there must be CCC.CCC.CCC.CCC
really strange, do you think it's an openser bug ?
The strangest thing is that I just copied openser configs from a
working system (Openser + Asterisks) changing just ip addresses
OpenSER version is also the same...
[...]
# !! Nathelper
onreply_route[1] {
# NATed transaction ?
if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") {
fix_nated_contact();
#force_rtp_proxy();
^
Are you sure it is commented? I do not believe it because the ngrep
shows that the SDP of 200 Ok is rewritten.
Sure, I also tried to remove the line without solving the previous problem
Please read the Getting Started Turial from
onsip.org
carefully. It
describes how you handle NAT correctly and also describes how to do
NAT traversal for in-dialog messages, which is missing in your config.
I will.
Tnx again
Regards
Edoardo