I am having issue with call routing in certain situation. I am using drouting module and permission module for authentications.
Here is the trace * # Call comes from 192.168.1.11 to 192.168.1.11* U 192.168.1.11:5060 -> 192.168.1.13:5060 INVITE sip:[email protected]:5060;user=phone;transport=UDP;maddr=192.168.1.11 SIP/2.0. Record-Route: <sip:192.168.1.11;ftag=dc7-13c4-5db56e-28caa21e-5db56e;lr=on>. Via: SIP/2.0/UDP 192.168.1.11;branch=z9hG4bK2c7a.d24c40b5.0. v: SIP/2.0/UDP 65.243.172.245:5060 ;branch=z9hG4bK31c0ba996f394d817b1f3364936c1b88.4a620e1. Record-Route: <sip:65.243.172.245:5060;lr>. v: SIP/2.0/UDP 63.110.102.239:5060 ;branch=z9hG4bK4be83d023d1c5e705dbce69dc257428e.61c8be4a;received=63.110.102.239. record-route: <sip:63.110.102.239;lr>. Remote-Party-ID: <sip:[email protected]<sip%3a%[email protected]> >;screen=yes;party=calling;privacy=off. f: <sip:[email protected]:5060 ;user=phone>;tag=dc7-13c4-5db56e-28caa21e-5db56e. t: <sip:[email protected]:5060;user=phone>. i: a1d74c18905eadc713c45db56e6e0cb7e4a9d9a091cba8848-0096-6871. CSeq: 1 INVITE. Max-Forwards: 16. k: 100rel, replaces. allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK. v: SIP/2.0/UDP SCR9:5060;maddr=199.173.94.144;branch=z9hG4bK-5db56e-6e0cb7e4-3dc36a18;received=199.173.94.144. m: <sip:199.173.94.144:5060;transport=UDP>. c: application/SDP. l: 233. . v=0. o=PVG 1260705241820 1260705241820 IN IP4 199.173.77.34. s=-. p=+1 6135555555. c=IN IP4 199.173.77.34. t=0 0. m=audio 55632 RTP/AVP 18 0 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=fmtp:18 annexb=no. # U 192.168.1.13:5060 -> 192.168.1.11:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 192.168.1.11;branch=z9hG4bK2c7a.d24c40b5.0. v: SIP/2.0/UDP 65.243.172.245:5060 ;branch=z9hG4bK31c0ba996f394d817b1f3364936c1b88.4a620e1. v: SIP/2.0/UDP 63.110.102.239:5060 ;branch=z9hG4bK4be83d023d1c5e705dbce69dc257428e.61c8be4a;received=63.110.102.239. f: <sip:[email protected]:5060 ;user=phone>;tag=dc7-13c4-5db56e-28caa21e-5db56e. t: <sip:[email protected]:5060;user=phone>. i: a1d74c18905eadc713c45db56e6e0cb7e4a9d9a091cba8848-0096-6871. CSeq: 1 INVITE. v: SIP/2.0/UDP SCR9:5060;maddr=199.173.94.144;branch=z9hG4bK-5db56e-6e0cb7e4-3dc36a18;received=199.173.94.144. Server: OpenSIPS (1.6.0-notls (x86_64/linux)). Content-Length: 0. . * According to drouting rules, call should be going to 192.168.1.3, but somehow opensips is updating the maddress to the origination IP (192.168.1.11 in this case) and is sending calls to *the IP in maddr. in place to destination IP. # Sending invite to originating IP. *U 192.168.1.13:5060 -> 192.168.1.11:5060 * *INVITE sip:[email protected] <sip%[email protected]>;user=phone;transport=UDP;maddr=192.168.1.11 SIP/2.0.* Record-Route: <sip:192.168.1.13;lr=on;ftag=dc7-13c4-5db56e-28caa21e-5db56e>. Record-Route: <sip:192.168.1.11;ftag=dc7-13c4-5db56e-28caa21e-5db56e;lr=on>. Via: SIP/2.0/UDP 192.168.1.13;branch=z9hG4bK2c7a.cb3da61.0. Via: SIP/2.0/UDP 192.168.1.11;branch=z9hG4bK2c7a.d24c40b5.0. v: SIP/2.0/UDP 65.243.172.245:5060 ;branch=z9hG4bK31c0ba996f394d817b1f3364936c1b88.4a620e1. Record-Route: <sip:65.243.172.245:5060;lr>. v: SIP/2.0/UDP 63.110.102.239:5060 ;branch=z9hG4bK4be83d023d1c5e705dbce69dc257428e.61c8be4a;received=63.110.102.239. record-route: <sip:63.110.102.239;lr>. Remote-Party-ID: <sip:[email protected]<sip%3a%[email protected]> >;screen=yes;party=calling;privacy=off. f: <sip:[email protected]:5060 ;user=phone>;tag=dc7-13c4-5db56e-28caa21e-5db56e. t: <sip:[email protected]:5060;user=phone>. i: a1d74c18905eadc713c45db56e6e0cb7e4a9d9a091cba8848-0096-6871. CSeq: 1 INVITE. Max-Forwards: 15. k: 100rel, replaces. allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK. v: SIP/2.0/UDP SCR9:5060;maddr=199.173.94.144;branch=z9hG4bK-5db56e-6e0cb7e4-3dc36a18;received=199.173.94.144. m: <sip:199.173.94.144:5060;transport=UDP>. c: application/SDP. l: 233. . v=0. o=PVG 1260705241820 1260705241820 IN IP4 199.173.77.34. s=-. p=+1 6135555555. c=IN IP4 199.173.77.34. t=0 0. m=audio 55632 RTP/AVP 18 0 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=p # My Routing logic is quite simple. if (is_uri_host_local()) { # -- outbound to inbound route(12); } route[4] { #Routing to Public Network if (!do_routing("10","1")) { xlog("-- do_routing failed \n"); sl_send_reply("503", "Unable to load gateways"); exit ; } else { t_on_failure("1"); #<--- This will be where you load the nextgateway route(1); exit; }; } # End Route 4 route[12] { # From an External Domain -> inbound lookup("aliases"); if (!lookup("location")) { route(4); exit ; } route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } I am confused why opensip is seding call to maddr IP in place of IP in destination URI. Any help of link will be appreciated. Best, -Jai
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