Dear all, i'm having scenario where opensips 1.6.3 and asterisk is on the same host. Opensips binds on port 5060 and asterisk on 5061.
Opensips handles user registration, nat traversal and "redirect" callls to Asterisk PBX (configured as pstn gateway in address table and there is no 407 proxy auth req so they trust each other :)) . Asterisk then via sip trunk send calls to our provider. Rtp proxy for nat traversal also running on same machine started and opensips do not report any errors. ( started like rtpproxy -l public_ip_opensips_and* -s udp:127.0.0.1:7890 -F -u rtpproxy). RTP proxy use default udp range 10000-35000 and asterisk use 36000-65534 range. UA is X-lite and behind nat. Calls are getting connected but they drop after 30 sec. X-lite recive 200 ok from * and sends back ACK. But ACK not getting to Asterisk. Asterisk reports retransmit timeout error. I'm think that problem is like they say in sip-retransmit file - A SIP middlebox (SBC) that rewrites contact: headers so that we can't reach the other side with our reply or the ACK. - A badly configured SIP proxy that forgets to add record-route headers to make sure that signalling works. When X-lite is in lan where is opensips error do not exists. Calls are working perfect! Here is 200 ok that asterisk keep retransmiting: ^[[0KRetransmitting #2 (no NAT) to 192.168.1.42:5060: SIP/2.0 200 OK v: SIP/2.0/UDP 192.168.1.42;branch=z9hG4bKe31e.f8aa9df6.0;received=192.168.1.42 v: SIP/2.0/UDP PUBLIC_IP_OF_UA:59788;rport=59788;received=PUBLIC_IP_OF_UA;branch=z9hG4bK-d8754z-c754d54c1f044c74-1---d8754z- Record-Route: <sip:192.168.1.42;lr=on> f: "tommy2" <sip:6557181...@public_ip_of_opensips_and*>;tag=740b0d14 t: "7890100" <sip:7890...@8public_ip_of_opensips_and*>;tag=as57556733 i: YjNjZThlNjljMjk2ODE5MmU1NDNiNTJhMTY5ZDg2MWQ. CSeq: 2 INVITE Server: Asterisk PBX ^[[0Kllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer m: <sip:[email protected]:5061> c: application/sdp l: 261 In opensips.cfg file i have this 2 section that involves ACK: if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after # a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction -> # ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } and : # preloaded route checking if (loose_route()) { xlog("L_ERR", "Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]"); if (!is_method("ACK")) sl_send_reply("403","Preload Route denied"); append_hf("P-hint: rr-enforced\r\n"); route(1); } i'm sending calls this way : route[4] { #---- PSTN route ----# rewritehostport("192.168.1.42:5061"); route(1); exit; } I hope there is solution for this and what to thanks in advanced anyone who try to help . Please help!!!! Cheers T0mmy -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-do-not-route-ACK-to-Asterisk-tp5746867p5746867.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
