Hi all, I'm trying to configure my OpenSIPS according to doc, with DR module. The issue happen when I send a call (from a softphone), in the second leg of the call I see several INVITES to TermGw.
I'm new in OpenSIPS ( I'm experienced with Nextone ), I could do working OpenSIPS routing just through opensips.cfg, but I need to have thousand of routes, so keep updated that file would be a heavy task. Please your help to know what's the mistake, where I'm wrong ? What's it is missed or omitted ?? Or please give me a working example !! With that I have enough. When I find the working template I want to work in scripting tools in order to handle thousand of routes easily, I'd share that with the community. Softphone IP is 192.168.0.18 OpenSIPS IP is 192.168.0.112 TermGw is 192.168.0.250 (Asterisk) Here the tshark trace: 6.833804 192.168.0.18 -> 192.168.0.112 SIP/SDP Request: INVITE sip:[email protected], with session descript ion 6.835459 192.168.0.112 -> 192.168.0.18 SIP Status: 100 Giving a try 6.835657 192.168.0.112 -> 192.168.0.250 SIP/SDP Request: INVITE sip:[email protected], with session descriptio n 6.836579 192.168.0.250 -> 192.168.0.112 SIP Status: 100 Trying 6.839673 192.168.0.112 -> 192.168.0.250 SIP/SDP Request: INVITE sip:[email protected], with session descriptio n 6.840750 192.168.0.250 -> 192.168.0.112 SIP Status: 100 Trying 6.844255 192.168.0.112 -> 192.168.0.250 SIP/SDP Request: INVITE sip:[email protected], with session descriptio n 6.844842 192.168.0.250 -> 192.168.0.112 SIP Status: 100 Trying 6.847648 192.168.0.112 -> 192.168.0.250 SIP/SDP Request: INVITE sip:[email protected], with session descriptio n 6.848220 192.168.0.250 -> 192.168.0.112 SIP Status: 100 Trying 6.851395 192.168.0.112 -> 192.168.0.250 SIP/SDP Request: INVITE sip:[email protected], with session descriptio n 6.852144 192.168.0.250 -> 192.168.0.112 SIP Status: 100 Trying 6.964467 192.168.0.250 -> 192.168.0.112 SIP/SDP Status: 183 Session Progress, with session description 6.965185 192.168.0.112 -> 192.168.0.18 SIP/SDP Status: 183 Session Progress, with session description --- After several secs I hang the call... 12.852208 192.168.0.18 -> 192.168.0.112 SIP Request: CANCEL sip:[email protected] 12.854316 192.168.0.112 -> 192.168.0.18 SIP Status: 200 canceling 12.855668 192.168.0.112 -> 192.168.0.250 SIP Request: CANCEL sip:[email protected] 12.856157 192.168.0.250 -> 192.168.0.112 SIP Status: 487 Request Terminated 12.856179 192.168.0.250 -> 192.168.0.112 SIP Status: 200 OK 12.856308 192.168.0.112 -> 192.168.0.250 SIP Request: ACK sip:[email protected] 12.858402 192.168.0.112 -> 192.168.0.18 SIP Status: 487 Request Terminated 12.863562 192.168.0.18 -> 192.168.0.112 SIP Request: ACK sip:[email protected] These are my data config in the database: * dr_gateways "gwid" "type" "address" "strip" "pri_prefix" "attrs" "probe_mode" "description" "2" "2" "192.168.0.250" "4" "9" null "0" "mygw1" * dr_groups "id" "username" "domain" "groupid" "description" "1" "9000" "192.168.0.112" "1" "edeww" "2" "1003" "192.168.0.112" "1" "Fer" * dr_gw_lists "id" "gwlist" "description" "1" "2" "GatewaysLocales" * dr_rules "ruleid" "groupid" "prefix" "timerec" "priority" "routeid" "gwlist" "description" "1" "1" "4712" "" "1" "4" "2" "MyFirstRoute" "2" "2" "4712" "" "1" "4" "2" "RouteFer" * subscriber "id" "username" "domain" "password" "email_address" "ha1" "ha1b" "rpid" "6" "1003" "sip.rudy-test.com" "12345" "[email protected]" "" "" null "5" "9000" "sip.rudy-test.com" "1234" "[email protected]" "" "" null Config in opensips.cfg: route{ # Do Accounting on INVITE, CANCEL AND BYE setflag(2); setflag(3); setflag(4); if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); } if (uri==myself) { if (method=="REGISTER") { setflag(6); force_rport(); # Uncomment this if you want to use digest authentication if (!www_authorize("192.168.0.112", "subscriber")) { www_challenge("192.168.0.112", "0"); exit; }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; if (uri=~"^sip:47...@*" ) { route(4); exit; } } route[1] { # for INVITEs enable some additional helper routes if (is_method("INVITE")) { # t_on_branch("2"); t_on_reply("2"); # t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); }; exit; } route[4] { #---- pSTN route ----# if(!do_routing()){ send_reply("503", "No rules found matching the given URI prefix "); exit; } # flag 10 flag the transaction to handle the failure route #setflag(10); route(1); } _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
