[OpenSIPS-Users] Slight problem routing 100s and 183s

[Nick Khamis]

Bogdan,

I see how busy you are with OpenSIPS so I will make it count.
Yes OpenSIP-Out is the new box that we have put in place to:

Bellow is a quick network diagram. The issue we are experiencing is that
the 100s, 183s and 200s
that come back from the carrier do not get processed or even responded to
by OpenSIPS-In.
The complete sip trace for OpenSIPS-In can be found at "
http://pastebin.com/iGeWsc40".
I did not include anything for "OUT" since it is performing as expected.

Some things to notice are the changed CallID. This is done by asterisk
(192.168.2.10):

Initial: Call-ID: 4737d441-5fb15ea7-7142c0d8 at 192.168.2.11.
Modified: Call-ID: 1fbe6fb90553da7c52d72b60076030f5 at 192.168.2.10:5060.

And the vanishing of RR: Record-Route: <sip:192.168.2.5;lr;did=b82.
180aabc6>.
This is also due to asterisk's recreation of the initial INVITE.

When it comes to network appliances, this is the last piece of the pie.