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@192.168.2.11. Modified: Call-ID: 1fbe6fb90553da7c52d72b60076030f5@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. >From now on it's mainly business logic, which should be less of a learning curve for us!!! I decided to post my problem online with example values, so it would hopefully help someone in the future. Kind Regards, Nick. [image: network.jpg]<https://mail.google.com/mail/ca/?ui=2&ik=e9f48992ab&view=att&th=13eb42dafefa444e&attid=0.1&disp=inline&realattid=f_hgttk2a11&safe=1&zw>
<<network.jpg>>
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