Hi, I'm using OpenSIPS in a test environment with an OSP server to provide authorization for calls. The setup is 2 phones and an opensips sip proxy. SIP proxy has an IP of 192.168.1.2 Testphone has an IP of 192.168.1.4 and number 1202 Testphone 2 has an IP of 192.168.1.5 and number 1203 With no accounts setup on the OSP server when I try and make a call the sip proxy returns a 503 because the device is not authorized by the OSP server. Now I have added devices to the OSP server and am trying to make a call from 1203 to 1202 I'm getting 404's. When looking at call detail records on the OSP server there are four calls there even though I only placed one. They all have sources of 192.168.1.5 and destinations of 1202 except one that has a destination of 192.168.1.4 which is the IP of the phone I want to reach. The output in the opensips logfile of the routing logic is shown below:
----ROUTE: Route IN - M=INVITE RURI=sip:[email protected] F=sip:[email protected] T=sip:[email protected] IP=192.168.1.5 [email protected] ----ROUTE: Processing INVITE OSP authorization validation logic Without OSP token, apply different authentication strategy Go ahead, everyone is welcomed ----ROUTE: Authentication passed ----ROUTE: Use OSP to get routing OSP authorization and routing logic Requesting OSP authorization and routing INFO:osp:ospRequestRouting: request auth and routing for: service_type '0' source '[192.168.1.2]:5060' source_dev '[192.168.1.5]:5060' source_networkid '' calling 'testphone2' called '1202' preferred '' nprn '' npcic '' npdi '0' spid '' ocn '' spn '' altspn '' mcc '' mnc '' diversion_user '' diversion_host '' call_id '[email protected]' dest_count '5' INFO:osp:ospLoadRoutes: get destination '0': valid after '2010-04-15T10:47:06Z' valid until '2010-04-15T10:57:06Z' time limit '14400' seconds call id '[email protected]' calling 'testphone2' called '1202' host '[192.168.1.5]' nprn '' npcic '' npdi '0' spid '' ocn '' spn '' altspn '' mcc '' mnc '' supported '1' network id '' token size '0' INFO:osp:ospLoadRoutes: get destination '1': valid after '2010-04-15T10:47:06Z' valid until '2010-04-15T10:57:06Z' time limit '14400' seconds call id '[email protected]' calling 'testphone2' called '1202' host '[192.168.1.4]' nprn '' npcic '' npdi '0' spid '' ocn '' spn '' altspn '' mcc '' mnc '' supported '1' network id '' token size '0' INFO:osp:ospRequestRouting: there are '2' OSP routes, call_id '[email protected]' Response received Try the 1st route Prepare route specific OSP information INFO:osp:ospPrepareDestination: prepare route to URI 'sip:[email protected]' for call_id '[email protected]' transaction_id '5460314288097329217' ----ROUTE: Route OUT Try the next route Prepare route specific OSP information INFO:osp:ospPrepareDestination: prepare route to URI 'sip:[email protected]' for call_id '[email protected]' transaction_id '5460314288097329217' Try the next route ----ROUTE: All destinations attempted for call ID '[email protected]'. Call cannot be completed. INFO:osp:ospReportOrigSetupUsage: report orig setup for call_id '[email protected]' transaction_id '5460314288097329217' ----ROUTE: Route IN - M=ACK RURI=sip:[email protected] F=sip:[email protected] T=sip:[email protected] IP=192.168.1.5 [email protected] ----ROUTE: Processing ACK ----ROUTE: Relay E2E ACK ----ROUTE: Route OUT I cant seem to make sense of whats going on, can anyone see what the problem is here? Thanks -- View this message in context: http://n2.nabble.com/Call-not-being-routed-correctly-with-OSP-module-tp4906928p4906928.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
