Hi all, can anyone help me to find out what is wrong with my setup, i have an asterisk behind a kamailio, kamailio is proxying all packages to the outside.
when the call is bridge it gets disconnected after a few seconds, it seems that our voip carrier is sending a bye because we didn't answer to their 200 ok propperly, but as the trace shows we did only that kamailio is answering to the contact header ip not the ip that is sending the ok.
any help is apreciated .
thanks.
my setup
request_route {
if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }
if(is_method("OPTIONS")) { # send reply for each options request sl_send_reply("200", "ok"); exit(); }
if(method=="BYE") { #Account BYE transactions
};
if (method=="CANCEL") { if (t_check_trans()) t_relay();
exit; };
if (loose_route()) {
t_relay(); exit; }
if (is_method("INVITE")) {
record_route();
} f (!t_relay_to_udp("3.1.1.1", "5060")) { sl_reply_error(); exit; }; exit };
here is a trace to a call made to a hotel. i had changed the real ips for obvious reasons. thanks.
asterisk ip 1.1.1.1 kamailio internal 1.1.1.2 kamailio external 2.0.0.1 Voip Carrier 3.1.1.1 voip contact ip 3.1.1.2
U 2013/10/23 17:26:03.920163 1.1.1.1:5060 -> 1.1.1.2:5060 INVITE sip:23276341079@2.0.0.1 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport. Max-Forwards: 70. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. To: sip:23276341079@2.0.0.1. Contact: sip:+19812457865@1.1.1.1:5060. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 1.8.15-cert2. Date: Wed, 23 Oct 2013 21:26:46 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Privacy: off. P-Asserted-Identity: sip:+19812457865@1.1.1.1. Cisco-Guid: 25655507-3591552378-379709 Content-Type: application/sdp. Content-Length: 333. . v=0. o=root 519803789 519803789 IN IP4 1.1.1.1. s=Asterisk PBX 1.8.15-cert2. c=IN IP4 1.1.1.1. t=0 0. m=audio 49926 RTP/AVP 0 18 3 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
U 2013/10/23 17:26:03.921355 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. To: sip:23276341079@2.0.0.1. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Server: kamailio (4.0.4 (x86_64/linux)). Content-Length: 0. .
U 2013/10/23 17:26:03.921544 1.1.1.2:5060 -> 3.1.1.1:5060 INVITE sip:76890723276341079@3.1.1.1:5060 SIP/2.0. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Max-Forwards: 16. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. To: sip:76890723276341079@3.1.1.1. Contact: sip:+19812457865@1.1.1.1:5060. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 1.8.15-cert2. Date: Wed, 23 Oct 2013 21:26:46 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Privacy: off. P-Asserted-Identity: sip:+19812457865@1.1.1.1. Cisco-Guid: 25655507-3591552378-379709 Content-Type: application/sdp. Content-Length: 333. . v=0. o=root 519803789 519803789 IN IP4 1.1.1.1. s=Asterisk PBX 1.8.15-cert2. c=IN IP4 1.1.1.1. t=0 0. m=audio 49926 RTP/AVP 0 18 3 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
U 2013/10/23 17:26:03.955394 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. To: sip:76890723276341079@3.1.1.1. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Server: gProxy (1.8.3 (i386/Linux)). Content-Length: 0. .
U 2013/10/23 17:26:04.424330 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: sip:76890723276341079@3.1.1.1;tag=3591552407-393967. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: sip:76890723276341079@3.1.1.2:5060. Call-Info: sip:3.1.1.2;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
U 2013/10/23 17:26:04.424521 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: sip:76890723276341079@3.1.1.1;tag=3591552407-393967. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: sip:76890723276341079@3.1.1.2:5060. Call-Info: sip:3.1.1.2;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
U 2013/10/23 17:26:16.846067 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: sip:76890723276341079@3.1.1.1;tag=3591552407-393967. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: sip:76890723276341079@3.1.1.2:5060. Call-Info: sip:3.1.1.2;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
U 2013/10/23 17:26:16.846201 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: sip:76890723276341079@3.1.1.1;tag=3591552407-393967. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: sip:76890723276341079@3.1.1.2:5060. Call-Info: sip:3.1.1.2;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
T 2013/10/23 17:26:16.846287 1.1.1.2:55305 -> 10.0.3.54:3306 [AP] .....insert into acc (method,from_tag,to_tag,callid,sip_code,sip_reason,time,from_uri,to_uri,kekuintid,type_call,dst_ip,carriercode,callmode ) values ('INVITE','as4bc322e9','3591552407-393967',' 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060','200','OK','2013-10-23 17:26:16','sip:+19812457865@1.1.1.1','sip:23276341079@2.0.0.1 ','+19812457865','1.1.1.1','sip:76890723276341079@3.1.1.1:5060',' sip:23276341079@2.0.0.1','OUT')
U 2013/10/23 17:26:16.847421 1.1.1.1:5060 -> 1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK05b1c5df;rport. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9,sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Max-Forwards: 70. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. To: sip:23276341079@2.0.0.1;tag=3591552407-393967. Contact: sip:+19812457865@1.1.1.1:5060. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. .
U 2013/10/23 17:26:16.847651 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK05b1c5df;rport=5060. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9,sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Max-Forwards: 16. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. To: sip:76890723276341079@3.1.1.2;tag=3591552407-393967. Contact: sip:+19812457865@1.1.1.1:5060. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. .
U 2013/10/23 17:26:17.346094 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: sip:76890723276341079@3.1.1.1;tag=3591552407-393967. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: sip:76890723276341079@3.1.1.2:5060. Call-Info: sip:3.1.1.2;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
U 2013/10/23 17:26:17.346262 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: sip:76890723276341079@3.1.1.1;tag=3591552407-393967. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: sip:76890723276341079@3.1.1.2:5060. Call-Info: sip:3.1.1.2;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
U 2013/10/23 17:26:17.349001 1.1.1.1:5060 -> 1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK675a9141;rport. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9,sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Max-Forwards: 70. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. To: sip:23276341079@2.0.0.1;tag=3591552407-393967. Contact: sip:+19812457865@1.1.1.1:5060. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. .
U 2013/10/23 17:26:17.349223 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK675a9141;rport=5060. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9,sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Max-Forwards: 16. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. To: sip:76890723276341079@3.1.1.2;tag=3591552407-393967. Contact: sip:+19812457865@1.1.1.1:5060. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. .
U 2013/10/23 17:26:18.347584 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: sip:76890723276341079@3.1.1.1;tag=3591552407-393967. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: sip:76890723276341079@3.1.1.2:5060. Call-Info: sip:3.1.1.2;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
U 2013/10/23 17:26:18.347767 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: sip:76890723276341079@3.1.1.1;tag=3591552407-393967. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: sip:76890723276341079@3.1.1.2:5060. Call-Info: sip:3.1.1.2;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
U 2013/10/23 17:26:18.348867 1.1.1.1:5060 -> 1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK3fc3c548;rport. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9,sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Max-Forwards: 70. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. To: sip:23276341079@2.0.0.1;tag=3591552407-393967. Contact: sip:+19812457865@1.1.1.1:5060. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. .
U 2013/10/23 17:26:18.349133 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK3fc3c548;rport=5060. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9,sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Max-Forwards: 16. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. To: sip:76890723276341079@3.1.1.2;tag=3591552407-393967. Contact: sip:+19812457865@1.1.1.1:5060. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. .
U 2013/10/23 17:26:20.352624 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: sip:76890723276341079@3.1.1.1;tag=3591552407-393967. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: sip:76890723276341079@3.1.1.2:5060. Call-Info: sip:3.1.1.2;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
U 2013/10/23 17:26:20.353056 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: sip:76890723276341079@3.1.1.1;tag=3591552407-393967. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: sip:76890723276341079@3.1.1.2:5060. Call-Info: sip:3.1.1.2;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
U 2013/10/23 17:26:20.354026 1.1.1.1:5060 -> 1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9,sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Max-Forwards: 70. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. To: sip:23276341079@2.0.0.1;tag=3591552407-393967. Contact: sip:+19812457865@1.1.1.1:5060. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. .
U 2013/10/23 17:26:20.354248 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport=5060. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9,sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Max-Forwards: 16. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. To: sip:76890723276341079@3.1.1.2;tag=3591552407-393967. Contact: sip:+19812457865@1.1.1.1:5060. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. .
U 2013/10/23 17:26:36.355580 3.1.1.1:5060 -> 1.1.1.2:5060 BYE sip:+19812457865@1.1.1.1:5060 SIP/2.0. Max-Forwards: 69. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. From: sip:76890723276341079@3.1.1.1;tag=3591552407-393967. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 2 BYE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Via: SIP/2.0/UDP 3.1.1.1:5060;branch=z9hG4bKce8a.db93afa3.0. Via: SIP/2.0/UDP 3.1.1.2:5060 ;branch=z9hG4bK96003ecbb11f5deaf6014235140e6952. Contact: sip:76890723276341079@3.1.1.2:5060. Content-Length: 0. .
U 2013/10/23 17:26:36.355995 1.1.1.2:5060 -> 1.1.1.1:5060 BYE sip:+19812457865@1.1.1.1:5060 SIP/2.0. Max-Forwards: 16. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: "+19812457865" sip:176822213@1.1.1.1;tag=as4bc322e9. From: 0 sip:079@3.1.1.1;tag=3591552407-393967. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 2 BYE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKce8a.52d22d63.0. Via: SIP/2.0/UDP 3.1.1.1:5060;branch=z9hG4bKce8a.db93afa3.0. Via: SIP/2.0/UDP 3.1.1.2:5060 ;branch=z9hG4bK96003ecbb11f5deaf6014235140e6952. Contact: sip:76890723276341079@3.1.1.2:5060. Content-Length: 0. .
Hello,
the config file is not complete, you don't pass the parameters. The ngrep trace that kamailio is forwarding the invite from the private IP but the route header is having the public IP.
You have to set debug=3 in your config and send the log messages here to see what is executed.
As a guess hit - if kamailio is listening on a private ip, being behind a port forwarding nat firewall, you may consider:
listen=udp:privateip:5060 advertise publicip:5060
See Core Cookbook from the kamailio.org wiki for more details about the above parameter.
Cheers, Daniel
On 10/24/13 12:01 AM, anfecora wrote:
Hi all, can anyone help me to find out what is wrong with my setup, i have an asterisk behind a kamailio, kamailio is proxying all packages to the outside.
when the call is bridge it gets disconnected after a few seconds, it seems that our voip carrier is sending a bye because we didn't answer to their 200 ok propperly, but as the trace shows we did only that kamailio is answering to the contact header ip not the ip that is sending the ok.
any help is apreciated .
thanks.
my setup
request_route {
if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS")) { # send reply for each options request sl_send_reply("200", "ok"); exit(); }if(method=="BYE") { #Account BYE transactions
};
if (method=="CANCEL") { if (t_check_trans()) t_relay();
exit; };
if (loose_route()) {
t_relay(); exit; }
if (is_method("INVITE")) {
record_route(); }f (!t_relay_to_udp("3.1.1.1", "5060")) { sl_reply_error(); exit; }; exit };
here is a trace to a call made to a hotel. i had changed the real ips for obvious reasons. thanks.
asterisk ip 1.1.1.1 kamailio internal 1.1.1.2 kamailio external 2.0.0.1 Voip Carrier 3.1.1.1 voip contact ip 3.1.1.2
U 2013/10/23 17:26:03.920163 1.1.1.1:5060 http://1.1.1.1:5060 -> 1.1.1.2:5060 http://1.1.1.2:5060 INVITE sip:23276341079@2.0.0.1 mailto:sip%3A23276341079@2.0.0.1 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1 mailto:sip%3A%2B19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1 mailto:sip%3A23276341079@2.0.0.1>. Contact: <sip:+19812457865@1.1.1.1:5060 http://sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060 http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 1.8.15-cert2. Date: Wed, 23 Oct 2013 21:26:46 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Privacy: off. P-Asserted-Identity: sip:+19812457865@1.1.1.1 mailto:sip%3A%2B19812457865@1.1.1.1. Cisco-Guid: 25655507-3591552378-379709 Content-Type: application/sdp. Content-Length: 333. . v=0. o=root 519803789 519803789 IN IP4 1.1.1.1. s=Asterisk PBX 1.8.15-cert2. c=IN IP4 1.1.1.1. t=0 0. m=audio 49926 RTP/AVP 0 18 3 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
U 2013/10/23 17:26:03.921355 1.1.1.2:5060 http://1.1.1.2:5060 -> 1.1.1.1:5060 http://1.1.1.1:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. From: "+19812457865" <sip:+19812457865@1.1.1.1 mailto:sip%3A%2B19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1 mailto:sip%3A23276341079@2.0.0.1>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060 http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Server: kamailio (4.0.4 (x86_64/linux)). Content-Length: 0. .
U 2013/10/23 17:26:03.921544 1.1.1.2:5060 http://1.1.1.2:5060 -> 3.1.1.1:5060 http://3.1.1.1:5060 INVITE sip:76890723276341079@3.1.1.1:5060 http://sip:76890723276341079@3.1.1.1:5060 SIP/2.0. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1 mailto:sip%3A%2B19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.1 mailto:sip%3A76890723276341079@3.1.1.1>. Contact: <sip:+19812457865@1.1.1.1:5060 http://sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060 http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 1.8.15-cert2. Date: Wed, 23 Oct 2013 21:26:46 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Privacy: off. P-Asserted-Identity: sip:+19812457865@1.1.1.1 mailto:sip%3A%2B19812457865@1.1.1.1. Cisco-Guid: 25655507-3591552378-379709 Content-Type: application/sdp. Content-Length: 333. . v=0. o=root 519803789 519803789 IN IP4 1.1.1.1. s=Asterisk PBX 1.8.15-cert2. c=IN IP4 1.1.1.1. t=0 0. m=audio 49926 RTP/AVP 0 18 3 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
U 2013/10/23 17:26:03.955394 3.1.1.1:5060 http://3.1.1.1:5060 -> 1.1.1.2:5060 http://1.1.1.2:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. From: "+19812457865" <sip:+19812457865@1.1.1.1 mailto:sip%3A%2B19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.1 mailto:sip%3A76890723276341079@3.1.1.1>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060 http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Server: gProxy (1.8.3 (i386/Linux)). Content-Length: 0. .
U 2013/10/23 17:26:04.424330 3.1.1.1:5060 http://3.1.1.1:5060 -> 1.1.1.2:5060 http://1.1.1.2:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: <sip:76890723276341079@3.1.1.1 mailto:sip%3A76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1 mailto:sip%3A%2B19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060 http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060 http://sip:76890723276341079@3.1.1.2:5060>. Call-Info: sip:3.1.1.2;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
U 2013/10/23 17:26:04.424521 1.1.1.2:5060 http://1.1.1.2:5060 -> 1.1.1.1:5060 http://1.1.1.1:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: <sip:76890723276341079@3.1.1.1 mailto:sip%3A76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1 mailto:sip%3A%2B19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060 http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060 http://sip:76890723276341079@3.1.1.2:5060>. Call-Info: sip:3.1.1.2;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
U 2013/10/23 17:26:16.846067 3.1.1.1:5060 http://3.1.1.1:5060 -> 1.1.1.2:5060 http://1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: <sip:76890723276341079@3.1.1.1 mailto:sip%3A76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1 mailto:sip%3A%2B19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060 http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060 http://sip:76890723276341079@3.1.1.2:5060>. Call-Info: sip:3.1.1.2;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
U 2013/10/23 17:26:16.846201 1.1.1.2:5060 http://1.1.1.2:5060 -> 1.1.1.1:5060 http://1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: <sip:76890723276341079@3.1.1.1 mailto:sip%3A76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1 mailto:sip%3A%2B19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060 http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060 http://sip:76890723276341079@3.1.1.2:5060>. Call-Info: sip:3.1.1.2;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
T 2013/10/23 17:26:16.846287 1.1.1.2:55305 http://1.1.1.2:55305 -> 10.0.3.54:3306 http://10.0.3.54:3306 [AP] .....insert into acc (method,from_tag,to_tag,callid,sip_code,sip_reason,time,from_uri,to_uri,kekuintid,type_call,dst_ip,carriercode,callmode ) values ('INVITE','as4bc322e9','3591552407-393967','7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060 http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060','200','OK','2013-10-23 17:26:16','sip:+19812457865@1.1.1.1 mailto:sip%3A%2B19812457865@1.1.1.1','sip:23276341079@2.0.0.1 mailto:sip%3A23276341079@2.0.0.1','+19812457865','1.1.1.1','sip:76890723276341079@3.1.1.1:5060 http://sip:76890723276341079@3.1.1.1:5060','sip:23276341079@2.0.0.1 mailto:sip%3A23276341079@2.0.0.1','OUT')
U 2013/10/23 17:26:16.847421 1.1.1.1:5060 http://1.1.1.1:5060 -> 1.1.1.2:5060 http://1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 http://sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK05b1c5df;rport. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9,sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1 mailto:sip%3A%2B19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1 mailto:sip%3A23276341079@2.0.0.1>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060 http://sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060 http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. .
U 2013/10/23 17:26:16.847651 1.1.1.2:5060 http://1.1.1.2:5060 -> 3.1.1.2:5060 http://3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 http://sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK05b1c5df;rport=5060. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9,sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1 mailto:sip%3A%2B19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2 mailto:sip%3A76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060 http://sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060 http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. .
U 2013/10/23 17:26:17.346094 3.1.1.1:5060 http://3.1.1.1:5060 -> 1.1.1.2:5060 http://1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: <sip:76890723276341079@3.1.1.1 mailto:sip%3A76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1 mailto:sip%3A%2B19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060 http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060 http://sip:76890723276341079@3.1.1.2:5060>. Call-Info: sip:3.1.1.2;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
U 2013/10/23 17:26:17.346262 1.1.1.2:5060 http://1.1.1.2:5060 -> 1.1.1.1:5060 http://1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: <sip:76890723276341079@3.1.1.1 mailto:sip%3A76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1 mailto:sip%3A%2B19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060 http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060 http://sip:76890723276341079@3.1.1.2:5060>. Call-Info: sip:3.1.1.2;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
U 2013/10/23 17:26:17.349001 1.1.1.1:5060 http://1.1.1.1:5060 -> 1.1.1.2:5060 http://1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 http://sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK675a9141;rport. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9,sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1 mailto:sip%3A%2B19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1 mailto:sip%3A23276341079@2.0.0.1>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060 http://sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060 http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. .
U 2013/10/23 17:26:17.349223 1.1.1.2:5060 http://1.1.1.2:5060 -> 3.1.1.2:5060 http://3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 http://sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK675a9141;rport=5060. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9,sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1 mailto:sip%3A%2B19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2 mailto:sip%3A76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060 http://sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060 http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. .
U 2013/10/23 17:26:18.347584 3.1.1.1:5060 http://3.1.1.1:5060 -> 1.1.1.2:5060 http://1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: <sip:76890723276341079@3.1.1.1 mailto:sip%3A76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1 mailto:sip%3A%2B19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060 http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060 http://sip:76890723276341079@3.1.1.2:5060>. Call-Info: sip:3.1.1.2;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
U 2013/10/23 17:26:18.347767 1.1.1.2:5060 http://1.1.1.2:5060 -> 1.1.1.1:5060 http://1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: <sip:76890723276341079@3.1.1.1 mailto:sip%3A76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1 mailto:sip%3A%2B19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060 http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060 http://sip:76890723276341079@3.1.1.2:5060>. Call-Info: sip:3.1.1.2;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
U 2013/10/23 17:26:18.348867 1.1.1.1:5060 http://1.1.1.1:5060 -> 1.1.1.2:5060 http://1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 http://sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK3fc3c548;rport. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9,sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1 mailto:sip%3A%2B19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1 mailto:sip%3A23276341079@2.0.0.1>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060 http://sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060 http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. .
U 2013/10/23 17:26:18.349133 1.1.1.2:5060 http://1.1.1.2:5060 -> 3.1.1.2:5060 http://3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 http://sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK3fc3c548;rport=5060. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9,sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1 mailto:sip%3A%2B19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2 mailto:sip%3A76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060 http://sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060 http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. .
U 2013/10/23 17:26:20.352624 3.1.1.1:5060 http://3.1.1.1:5060 -> 1.1.1.2:5060 http://1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: <sip:76890723276341079@3.1.1.1 mailto:sip%3A76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1 mailto:sip%3A%2B19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060 http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060 http://sip:76890723276341079@3.1.1.2:5060>. Call-Info: sip:3.1.1.2;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
U 2013/10/23 17:26:20.353056 1.1.1.2:5060 http://1.1.1.2:5060 -> 1.1.1.1:5060 http://1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: <sip:76890723276341079@3.1.1.1 mailto:sip%3A76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1 mailto:sip%3A%2B19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060 http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060 http://sip:76890723276341079@3.1.1.2:5060>. Call-Info: sip:3.1.1.2;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
U 2013/10/23 17:26:20.354026 1.1.1.1:5060 http://1.1.1.1:5060 -> 1.1.1.2:5060 http://1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 http://sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9,sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1 mailto:sip%3A%2B19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1 mailto:sip%3A23276341079@2.0.0.1>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060 http://sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060 http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. .
U 2013/10/23 17:26:20.354248 1.1.1.2:5060 http://1.1.1.2:5060 -> 3.1.1.2:5060 http://3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 http://sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport=5060. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9,sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1 mailto:sip%3A%2B19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2 mailto:sip%3A76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060 http://sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060 http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. .
U 2013/10/23 17:26:36.355580 3.1.1.1:5060 http://3.1.1.1:5060 -> 1.1.1.2:5060 http://1.1.1.2:5060 BYE sip:+19812457865@1.1.1.1:5060 http://sip:+19812457865@1.1.1.1:5060 SIP/2.0. Max-Forwards: 69. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: "+19812457865" <sip:+19812457865@1.1.1.1 mailto:sip%3A%2B19812457865@1.1.1.1>;tag=as4bc322e9. From: <sip:76890723276341079@3.1.1.1 mailto:sip%3A76890723276341079@3.1.1.1>;tag=3591552407-393967. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060 http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 2 BYE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Via: SIP/2.0/UDP 3.1.1.1:5060;branch=z9hG4bKce8a.db93afa3.0. Via: SIP/2.0/UDP 3.1.1.2:5060;branch=z9hG4bK96003ecbb11f5deaf6014235140e6952. Contact: <sip:76890723276341079@3.1.1.2:5060 http://sip:76890723276341079@3.1.1.2:5060>. Content-Length: 0. .
U 2013/10/23 17:26:36.355995 1.1.1.2:5060 http://1.1.1.2:5060 -> 1.1.1.1:5060 http://1.1.1.1:5060 BYE sip:+19812457865@1.1.1.1:5060 http://sip:+19812457865@1.1.1.1:5060 SIP/2.0. Max-Forwards: 16. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: "+19812457865" <sip:176822213@1.1.1.1 mailto:sip%3A176822213@1.1.1.1>;tag=as4bc322e9. From: 0 <sip:079@3.1.1.1 mailto:sip%3A079@3.1.1.1>;tag=3591552407-393967. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060 http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 2 BYE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKce8a.52d22d63.0. Via: SIP/2.0/UDP 3.1.1.1:5060;branch=z9hG4bKce8a.db93afa3.0. Via: SIP/2.0/UDP 3.1.1.2:5060;branch=z9hG4bK96003ecbb11f5deaf6014235140e6952. Contact: <sip:76890723276341079@3.1.1.2:5060 http://sip:76890723276341079@3.1.1.2:5060>. Content-Length: 0. .
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Thank you , yes it is behind a private IP, and I am using advertise option,
i was able to solve my issue using modparam("rr", "enable_double_rr", 0)
and then resplying ack o a loose_route() now it seems to work fine.
thanks Andres Collazos,
On Tue, Oct 29, 2013 at 5:34 AM, Daniel-Constantin Mierla <miconda@gmail.com
wrote:
Hello,
the config file is not complete, you don't pass the parameters. The ngrep trace that kamailio is forwarding the invite from the private IP but the route header is having the public IP.
You have to set debug=3 in your config and send the log messages here to see what is executed.
As a guess hit - if kamailio is listening on a private ip, being behind a port forwarding nat firewall, you may consider:
listen=udp:privateip:5060 advertise publicip:5060
See Core Cookbook from the kamailio.org wiki for more details about the above parameter.
Cheers, Daniel
On 10/24/13 12:01 AM, anfecora wrote:
Hi all, can anyone help me to find out what is wrong with my setup, i have an asterisk behind a kamailio, kamailio is proxying all packages to the outside.
when the call is bridge it gets disconnected after a few seconds, it seems that our voip carrier is sending a bye because we didn't answer to their 200 ok propperly, but as the trace shows we did only that kamailio is answering to the contact header ip not the ip that is sending the ok.
any help is apreciated .
thanks.
my setup
request_route {
if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS")) { # send reply for each options request sl_send_reply("200", "ok"); exit(); } if(method=="BYE") {#Account BYE transactions
};
if (method=="CANCEL") { if (t_check_trans()) t_relay();
exit; };
if (loose_route()) {
t_relay(); exit; }
if (is_method("INVITE")) {
record_route(); }f (!t_relay_to_udp("3.1.1.1", "5060")) { sl_reply_error(); exit; }; exit };
here is a trace to a call made to a hotel. i had changed the real ips for obvious reasons. thanks.
asterisk ip 1.1.1.1 kamailio internal 1.1.1.2 kamailio external 2.0.0.1 Voip Carrier 3.1.1.1 voip contact ip 3.1.1.2
U 2013/10/23 17:26:03.920163 1.1.1.1:5060 -> 1.1.1.2:5060 INVITE sip:23276341079@2.0.0.1 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport. Max-Forwards: 70. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. To: sip:23276341079@2.0.0.1. Contact: sip:+19812457865@1.1.1.1:5060. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 1.8.15-cert2. Date: Wed, 23 Oct 2013 21:26:46 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Privacy: off. P-Asserted-Identity: sip:+19812457865@1.1.1.1. Cisco-Guid: 25655507-3591552378-379709 Content-Type: application/sdp. Content-Length: 333. . v=0. o=root 519803789 519803789 IN IP4 1.1.1.1. s=Asterisk PBX 1.8.15-cert2. c=IN IP4 1.1.1.1. t=0 0. m=audio 49926 RTP/AVP 0 18 3 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
U 2013/10/23 17:26:03.921355 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. To: sip:23276341079@2.0.0.1. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Server: kamailio (4.0.4 (x86_64/linux)). Content-Length: 0. .
U 2013/10/23 17:26:03.921544 1.1.1.2:5060 -> 3.1.1.1:5060 INVITE sip:76890723276341079@3.1.1.1:5060 SIP/2.0. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Max-Forwards: 16. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. To: sip:76890723276341079@3.1.1.1. Contact: sip:+19812457865@1.1.1.1:5060. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 1.8.15-cert2. Date: Wed, 23 Oct 2013 21:26:46 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Privacy: off. P-Asserted-Identity: sip:+19812457865@1.1.1.1. Cisco-Guid: 25655507-3591552378-379709 Content-Type: application/sdp. Content-Length: 333. . v=0. o=root 519803789 519803789 IN IP4 1.1.1.1. s=Asterisk PBX 1.8.15-cert2. c=IN IP4 1.1.1.1. t=0 0. m=audio 49926 RTP/AVP 0 18 3 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
U 2013/10/23 17:26:03.955394 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. To: sip:76890723276341079@3.1.1.1. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Server: gProxy (1.8.3 (i386/Linux)). Content-Length: 0. .
U 2013/10/23 17:26:04.424330 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: sip:76890723276341079@3.1.1.1;tag=3591552407-393967. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: sip:76890723276341079@3.1.1.2:5060. Call-Info: sip:3.1.1.2 ;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
U 2013/10/23 17:26:04.424521 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: sip:76890723276341079@3.1.1.1;tag=3591552407-393967. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: sip:76890723276341079@3.1.1.2:5060. Call-Info: sip:3.1.1.2 ;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
U 2013/10/23 17:26:16.846067 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: sip:76890723276341079@3.1.1.1;tag=3591552407-393967. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: sip:76890723276341079@3.1.1.2:5060. Call-Info: sip:3.1.1.2 ;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
U 2013/10/23 17:26:16.846201 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: sip:76890723276341079@3.1.1.1;tag=3591552407-393967. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: sip:76890723276341079@3.1.1.2:5060. Call-Info: sip:3.1.1.2 ;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
T 2013/10/23 17:26:16.846287 1.1.1.2:55305 -> 10.0.3.54:3306 [AP] .....insert into acc (method,from_tag,to_tag,callid,sip_code,sip_reason,time,from_uri,to_uri,kekuintid,type_call,dst_ip,carriercode,callmode ) values ('INVITE','as4bc322e9','3591552407-393967',' 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060','200','OK','2013-10-23 17:26:16','sip:+19812457865@1.1.1.1','sip:23276341079@2.0.0.1 ','+19812457865','1.1.1.1','sip:76890723276341079@3.1.1.1:5060',' sip:23276341079@2.0.0.1','OUT')
U 2013/10/23 17:26:16.847421 1.1.1.1:5060 -> 1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK05b1c5df;rport. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9, sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Max-Forwards: 70. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. To: sip:23276341079@2.0.0.1;tag=3591552407-393967. Contact: sip:+19812457865@1.1.1.1:5060. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. .
U 2013/10/23 17:26:16.847651 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK05b1c5df;rport=5060. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9, sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Max-Forwards: 16. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. To: sip:76890723276341079@3.1.1.2;tag=3591552407-393967. Contact: sip:+19812457865@1.1.1.1:5060. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. .
U 2013/10/23 17:26:17.346094 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: sip:76890723276341079@3.1.1.1;tag=3591552407-393967. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: sip:76890723276341079@3.1.1.2:5060. Call-Info: sip:3.1.1.2 ;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
U 2013/10/23 17:26:17.346262 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: sip:76890723276341079@3.1.1.1;tag=3591552407-393967. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: sip:76890723276341079@3.1.1.2:5060. Call-Info: sip:3.1.1.2 ;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
U 2013/10/23 17:26:17.349001 1.1.1.1:5060 -> 1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK675a9141;rport. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9, sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Max-Forwards: 70. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. To: sip:23276341079@2.0.0.1;tag=3591552407-393967. Contact: sip:+19812457865@1.1.1.1:5060. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. .
U 2013/10/23 17:26:17.349223 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK675a9141;rport=5060. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9, sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Max-Forwards: 16. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. To: sip:76890723276341079@3.1.1.2;tag=3591552407-393967. Contact: sip:+19812457865@1.1.1.1:5060. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. .
U 2013/10/23 17:26:18.347584 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: sip:76890723276341079@3.1.1.1;tag=3591552407-393967. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: sip:76890723276341079@3.1.1.2:5060. Call-Info: sip:3.1.1.2 ;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
U 2013/10/23 17:26:18.347767 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: sip:76890723276341079@3.1.1.1;tag=3591552407-393967. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: sip:76890723276341079@3.1.1.2:5060. Call-Info: sip:3.1.1.2 ;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
U 2013/10/23 17:26:18.348867 1.1.1.1:5060 -> 1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK3fc3c548;rport. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9, sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Max-Forwards: 70. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. To: sip:23276341079@2.0.0.1;tag=3591552407-393967. Contact: sip:+19812457865@1.1.1.1:5060. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. .
U 2013/10/23 17:26:18.349133 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK3fc3c548;rport=5060. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9, sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Max-Forwards: 16. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. To: sip:76890723276341079@3.1.1.2;tag=3591552407-393967. Contact: sip:+19812457865@1.1.1.1:5060. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. .
U 2013/10/23 17:26:20.352624 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: sip:76890723276341079@3.1.1.1;tag=3591552407-393967. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: sip:76890723276341079@3.1.1.2:5060. Call-Info: sip:3.1.1.2 ;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
U 2013/10/23 17:26:20.353056 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Record-Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: sip:76890723276341079@3.1.1.1;tag=3591552407-393967. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: sip:76890723276341079@3.1.1.2:5060. Call-Info: sip:3.1.1.2 ;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
U 2013/10/23 17:26:20.354026 1.1.1.1:5060 -> 1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9, sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Max-Forwards: 70. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. To: sip:23276341079@2.0.0.1;tag=3591552407-393967. Contact: sip:+19812457865@1.1.1.1:5060. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. .
U 2013/10/23 17:26:20.354248 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport=5060. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9, sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05. Max-Forwards: 16. From: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. To: sip:76890723276341079@3.1.1.2;tag=3591552407-393967. Contact: sip:+19812457865@1.1.1.1:5060. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. .
U 2013/10/23 17:26:36.355580 3.1.1.1:5060 -> 1.1.1.2:5060 BYE sip:+19812457865@1.1.1.1:5060 SIP/2.0. Max-Forwards: 69. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: "+19812457865" sip:+19812457865@1.1.1.1;tag=as4bc322e9. From: sip:76890723276341079@3.1.1.1;tag=3591552407-393967. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 2 BYE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Via: SIP/2.0/UDP 3.1.1.1:5060;branch=z9hG4bKce8a.db93afa3.0. Via: SIP/2.0/UDP 3.1.1.2:5060 ;branch=z9hG4bK96003ecbb11f5deaf6014235140e6952. Contact: sip:76890723276341079@3.1.1.2:5060. Content-Length: 0. .
U 2013/10/23 17:26:36.355995 1.1.1.2:5060 -> 1.1.1.1:5060 BYE sip:+19812457865@1.1.1.1:5060 SIP/2.0. Max-Forwards: 16. Route: sip:2.0.0.1;lr=on;ftag=as4bc322e9. To: "+19812457865" sip:176822213@1.1.1.1;tag=as4bc322e9. From: 0 sip:079@3.1.1.1;tag=3591552407-393967. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 2 BYE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKce8a.52d22d63.0. Via: SIP/2.0/UDP 3.1.1.1:5060;branch=z9hG4bKce8a.db93afa3.0. Via: SIP/2.0/UDP 3.1.1.2:5060 ;branch=z9hG4bK96003ecbb11f5deaf6014235140e6952. Contact: sip:76890723276341079@3.1.1.2:5060. Content-Length: 0. .
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Nov 25-28
- more details about Kamailio trainings at http://www.asipto.com -
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-
anfecora -
Daniel-Constantin Mierla