Hi Nick, I guess you simply have 2 calls in there.
The callid : 4737d441-5fb15ea7-7142c0d8@192.168.2.11 comes from .11(phone) goes to .5(opensips) and to .10 (asterisk) - this call is not picked up (there is only a trying from asterisk), so .11 fires a CANCEL which ends the call. I do not know what is about the replies with callid 1fbe6fb90553da7c52d72b60076030f5@192.168.2.10:5060 - they seems to belong to another call, involving a party with public IP, but I do not see the INVITE, just some replies. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 05/21/2013 08:19 PM, Nick Khamis wrote: > Bogdan I am so sorry!!! 192.168.2.11 is actually a UAC polycom phone. > The only asterisk box that is being used in the scenario right now is > 192.168.2.10, as seen in the traces. Please forgive me! :) > > N. > > On 5/21/13, Bogdan-Andrei Iancu <bog...@opensips.org> wrote: >> Hello Nick, >> >> To be honest, I'm a bit confused - looking at the trace, I see the >> INVITE comes from .11 (an aterisks), goes to .5 (opensipsIn) and then to >> .10 (another asterisk)....This does not match the network diagram .. >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> >> On 05/17/2013 11:30 PM, Nick Khamis wrote: >>> 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 >>> <mailto:4737d441-5fb15ea7-7142c0d8@192.168.2.11>. >>> Modified: Call-ID: 1fbe6fb90553da7c52d72b60076030f5@192.168.2.10:5060 >>> <http://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. >>> >>> 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> >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users@lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users