Den 30/06/2010 kl. 01.23 skrev Iñaki Baz Castillo: > 2010/6/29 Ole Kaas <o...@tet.dk>: >> Hi Klaus, >> >> I've mailed pcap dump to you directly for further inspection. > > Hi, it's much better if you capture a trace with "ngrep -Wbyline -t -q > port 5060" and paste it in a new mail by replacing your public IP's > with other values. Then all the people here could help you rather than > asking for private help to a specific member of the maillist. >
You are right. But maybe it was something (obvious) that could be resolved quickly and I could post an update here on the list. The original log was inadequate - Klaus was a great help, with suggestions to obtain new log. So here it is attached and anonymized with all ip addresses (and stuff) converted to private adresses. The Kamailio server is multi homed and the two networks are non-routable (I use rtpproxy to bridge media). Our Asterisk PBX is version 1.4.26.1 and the Asterisk Gateway is 1.6.1 (or 1.6.0 - cant remember and not under my control). Kamailio is version 3.0.0. Looking at the trace, it seems the problem starts with the ACK not being received by the Asterisk Gateway which then resends the OK. The OK is relayed back to the originating Asterisk PBX which seems to ignore the retransmission. In fact it seems that Kamailio is routing and relaying the sip packets correctly. However, it seems that the problem only exists between Asterisk and Kamailio. I have other pbx'es (3CX) connecting to Kamailio and I have no evidence that the problem happens with those. I have another trace where the call comes from one of the Asterisk Gateways and is routed back to one of the other Asterisk Gateways. The result is the same - the OK's are ignored by Asterisk. /Ole
input: call-fail3-pcap U 2010/06/30 09:39:54.415523 192.168.1.87:5060 -> 192.168.1.94:5060 INVITE sip:22222...@192.168.1.94 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.87:5060;branch=z9hG4bK0eaddfef;rport. From: "Sip User" <sip:11111...@192.168.1.87>;tag=as6b8682ff. To: <sip:22222...@192.168.1.94>. Contact: <sip:11111...@192.168.1.87>. Call-ID: 3bb6091225147e4a419fb5ca44e45...@192.168.1.87. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Date: Wed, 30 Jun 2010 07:39:54 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. X-accountcode: 1028707002. Content-Type: application/sdp. Content-Length: 264. . v=0. o=root 995 995 IN IP4 192.168.1.87. s=session. c=IN IP4 192.168.1.87. t=0 0. m=audio 16956 RTP/AVP 0 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 2010/06/30 09:39:54.418241 192.168.1.94:5060 -> 192.168.1.87:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.1.87:5060;branch=z9hG4bK0eaddfef;rport=5060. From: "Sip User" <sip:11111...@192.168.1.87>;tag=as6b8682ff. To: <sip:22222...@192.168.1.94>. Call-ID: 3bb6091225147e4a419fb5ca44e45...@192.168.1.87. CSeq: 102 INVITE. Server: kamailio (3.0.0 (x86_64/linux)). Content-Length: 0. Warning: 392 192.168.1.94:5060 "Noisy feedback tells: pid=17701 req_src_ip=192.168.1.87 req_src_port=5060 in_uri=sip:22222...@192.168.1.94 out_uri=sip:22222...@10.0.0.54 via_cnt==1". . U 2010/06/30 09:39:54.418526 10.10.10.154:5060 -> 10.0.0.54:5060 INVITE sip:22222...@10.0.0.54 SIP/2.0. Record-Route: <sip:10.10.10.154;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. Record-Route: <sip:192.168.1.94;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. Via: SIP/2.0/UDP 10.10.10.154;branch=z9hG4bK7e45.61f92b7.0. Via: SIP/2.0/UDP 192.168.1.87:5060;branch=z9hG4bK0eaddfef;rport=5060. From: "Sip User" <sip:11111...@192.168.1.87>;tag=as6b8682ff. To: <sip:22222...@192.168.1.94>. Contact: <sip:11111...@192.168.1.87>. Call-ID: 3bb6091225147e4a419fb5ca44e45...@192.168.1.87. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 69. Date: Wed, 30 Jun 2010 07:39:54 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Content-Type: application/sdp. Content-Length: 282. P-Asserted-Identity: "Sip User" <sip:11111...@domain.tld>. X-accountcode: 3208171. . v=0. o=root 995 995 IN IP4 10.10.10.154. s=session. c=IN IP4 10.10.10.154. t=0 0. m=audio 39400 RTP/AVP 0 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. a=nortpproxy:yes. U 2010/06/30 09:39:54.427337 10.0.0.54:5060 -> 10.10.10.154:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 10.10.10.154;branch=z9hG4bK7e45.61f92b7.0;received=10.10.10.154. Via: SIP/2.0/UDP 192.168.1.87:5060;branch=z9hG4bK0eaddfef;rport=5060. Record-Route: <sip:10.10.10.154;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. Record-Route: <sip:192.168.1.94;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. From: "Sip User" <sip:11111...@192.168.1.87>;tag=as6b8682ff. To: <sip:22222...@192.168.1.94>. Call-ID: 3bb6091225147e4a419fb5ca44e45...@192.168.1.87. CSeq: 102 INVITE. Server: Asterisk Gateway. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Contact: <sip:22222...@10.0.0.54>. Content-Length: 0. . U 2010/06/30 09:39:54.730272 10.0.0.54:5060 -> 10.10.10.154:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 10.10.10.154;branch=z9hG4bK7e45.61f92b7.0;received=10.10.10.154. Via: SIP/2.0/UDP 192.168.1.87:5060;branch=z9hG4bK0eaddfef;rport=5060. Record-Route: <sip:10.10.10.154;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. Record-Route: <sip:192.168.1.94;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. From: "Sip User" <sip:11111...@192.168.1.87>;tag=as6b8682ff. To: <sip:22222...@192.168.1.94>;tag=as0736eaf3. Call-ID: 3bb6091225147e4a419fb5ca44e45...@192.168.1.87. CSeq: 102 INVITE. Server: Asterisk Gateway. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Contact: <sip:22222...@10.0.0.54>. Content-Length: 0. . U 2010/06/30 09:39:54.730376 10.0.0.54:5060 -> 10.10.10.154:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 10.10.10.154;branch=z9hG4bK7e45.61f92b7.0;received=10.10.10.154. Via: SIP/2.0/UDP 192.168.1.87:5060;branch=z9hG4bK0eaddfef;rport=5060. Record-Route: <sip:10.10.10.154;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. Record-Route: <sip:192.168.1.94;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. From: "Sip User" <sip:11111...@192.168.1.87>;tag=as6b8682ff. To: <sip:22222...@192.168.1.94>;tag=as0736eaf3. Call-ID: 3bb6091225147e4a419fb5ca44e45...@192.168.1.87. CSeq: 102 INVITE. Server: Asterisk Gateway. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Contact: <sip:22222...@10.0.0.54>. Content-Type: application/sdp. Content-Length: 253. . v=0. o=root 1451497710 1451497710 IN IP4 10.0.0.54. s=Asterisk Gateway. c=IN IP4 10.0.0.54. t=0 0. m=audio 19266 RTP/AVP 8 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2010/06/30 09:39:54.730507 192.168.1.94:5060 -> 192.168.1.87:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.1.87:5060;branch=z9hG4bK0eaddfef;rport=5060. Record-Route: <sip:10.10.10.154;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. Record-Route: <sip:192.168.1.94;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. From: "Sip User" <sip:11111...@192.168.1.87>;tag=as6b8682ff. To: <sip:22222...@192.168.1.94>;tag=as0736eaf3. Call-ID: 3bb6091225147e4a419fb5ca44e45...@192.168.1.87. CSeq: 102 INVITE. Server: Asterisk Gateway. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Contact: <sip:22222...@10.0.0.54>. Content-Length: 0. . U 2010/06/30 09:39:54.734451 192.168.1.94:5060 -> 192.168.1.87:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 192.168.1.87:5060;branch=z9hG4bK0eaddfef;rport=5060. Record-Route: <sip:10.10.10.154;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. Record-Route: <sip:192.168.1.94;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. From: "Sip User" <sip:11111...@192.168.1.87>;tag=as6b8682ff. To: <sip:22222...@192.168.1.94>;tag=as0736eaf3. Call-ID: 3bb6091225147e4a419fb5ca44e45...@192.168.1.87. CSeq: 102 INVITE. Server: Asterisk Gateway. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Contact: <sip:22222...@10.0.0.54>. Content-Type: application/sdp. Content-Length: 273. . v=0. o=root 1451497710 1451497710 IN IP4 192.168.1.94. s=Asterisk Gateway. c=IN IP4 192.168.1.94. t=0 0. m=audio 46066 RTP/AVP 8 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=nortpproxy:yes. U 2010/06/30 09:39:59.651623 10.0.0.54:5060 -> 10.10.10.154:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.10.10.154;branch=z9hG4bK7e45.61f92b7.0;received=10.10.10.154. Via: SIP/2.0/UDP 192.168.1.87:5060;branch=z9hG4bK0eaddfef;rport=5060. Record-Route: <sip:10.10.10.154;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. Record-Route: <sip:192.168.1.94;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. From: "Sip User" <sip:11111...@192.168.1.87>;tag=as6b8682ff. To: <sip:22222...@192.168.1.94>;tag=as0736eaf3. Call-ID: 3bb6091225147e4a419fb5ca44e45...@192.168.1.87. CSeq: 102 INVITE. Server: Asterisk Gateway. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Contact: <sip:22222...@10.0.0.54>. Content-Type: application/sdp. Content-Length: 253. . v=0. o=root 1451497710 1451497711 IN IP4 10.0.0.54. s=Asterisk Gateway. c=IN IP4 10.0.0.54. t=0 0. m=audio 19266 RTP/AVP 8 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2010/06/30 09:39:59.652362 192.168.1.94:5060 -> 192.168.1.87:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.87:5060;branch=z9hG4bK0eaddfef;rport=5060. Record-Route: <sip:10.10.10.154;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. Record-Route: <sip:192.168.1.94;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. From: "Sip User" <sip:11111...@192.168.1.87>;tag=as6b8682ff. To: <sip:22222...@192.168.1.94>;tag=as0736eaf3. Call-ID: 3bb6091225147e4a419fb5ca44e45...@192.168.1.87. CSeq: 102 INVITE. Server: Asterisk Gateway. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Contact: <sip:22222...@10.0.0.54>. Content-Type: application/sdp. Content-Length: 273. . v=0. o=root 1451497710 1451497711 IN IP4 192.168.1.94. s=Asterisk Gateway. c=IN IP4 192.168.1.94. t=0 0. m=audio 46066 RTP/AVP 8 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=nortpproxy:yes. U 2010/06/30 09:39:59.652548 192.168.1.87:5060 -> 192.168.1.94:5060 ACK sip:22222...@10.0.0.54 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.87:5060;branch=z9hG4bK140fcd7d;rport. Route: <sip:192.168.1.94;r2=on;lr=on;ftag=as6b8682ff;nat=yes>,<sip:10.10.10.154;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. From: "Sip User" <sip:11111...@192.168.1.87>;tag=as6b8682ff. To: <sip:22222...@192.168.1.94>;tag=as0736eaf3. Contact: <sip:11111...@192.168.1.87>. Call-ID: 3bb6091225147e4a419fb5ca44e45...@192.168.1.87. CSeq: 102 ACK. User-Agent: Asterisk PBX. Max-Forwards: 70. Content-Length: 0. . U 2010/06/30 09:39:59.653638 10.10.10.154:5060 -> 10.0.0.54:5060 ACK sip:22222...@10.0.0.54 SIP/2.0. Via: SIP/2.0/UDP 10.10.10.154;branch=0. Via: SIP/2.0/UDP 192.168.1.87:5060;branch=z9hG4bK140fcd7d;rport=5060. From: "Sip User" <sip:11111...@192.168.1.87>;tag=as6b8682ff. To: <sip:22222...@192.168.1.94>;tag=as0736eaf3. Contact: <sip:11111...@192.168.1.87>. Call-ID: 3bb6091225147e4a419fb5ca44e45...@192.168.1.87. CSeq: 102 ACK. User-Agent: Asterisk PBX. Max-Forwards: 69. Content-Length: 0. . U 2010/06/30 09:40:00.651072 10.0.0.54:5060 -> 10.10.10.154:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.10.10.154;branch=z9hG4bK7e45.61f92b7.0;received=10.10.10.154. Via: SIP/2.0/UDP 192.168.1.87:5060;branch=z9hG4bK0eaddfef;rport=5060. Record-Route: <sip:10.10.10.154;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. Record-Route: <sip:192.168.1.94;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. From: "Sip User" <sip:11111...@192.168.1.87>;tag=as6b8682ff. To: <sip:22222...@192.168.1.94>;tag=as0736eaf3. Call-ID: 3bb6091225147e4a419fb5ca44e45...@192.168.1.87. CSeq: 102 INVITE. Server: Asterisk Gateway. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Contact: <sip:22222...@10.0.0.54>. Content-Type: application/sdp. Content-Length: 253. . v=0. o=root 1451497710 1451497711 IN IP4 10.0.0.54. s=Asterisk Gateway. c=IN IP4 10.0.0.54. t=0 0. m=audio 19266 RTP/AVP 8 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2010/06/30 09:40:00.655994 192.168.1.94:5060 -> 192.168.1.87:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.87:5060;branch=z9hG4bK0eaddfef;rport=5060. Record-Route: <sip:10.10.10.154;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. Record-Route: <sip:192.168.1.94;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. From: "Sip User" <sip:11111...@192.168.1.87>;tag=as6b8682ff. To: <sip:22222...@192.168.1.94>;tag=as0736eaf3. Call-ID: 3bb6091225147e4a419fb5ca44e45...@192.168.1.87. CSeq: 102 INVITE. Server: Asterisk Gateway. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Contact: <sip:22222...@10.0.0.54>. Content-Type: application/sdp. Content-Length: 273. . v=0. o=root 1451497710 1451497711 IN IP4 192.168.1.94. s=Asterisk Gateway. c=IN IP4 192.168.1.94. t=0 0. m=audio 46066 RTP/AVP 8 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=nortpproxy:yes. U 2010/06/30 09:40:01.651104 10.0.0.54:5060 -> 10.10.10.154:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.10.10.154;branch=z9hG4bK7e45.61f92b7.0;received=10.10.10.154. Via: SIP/2.0/UDP 192.168.1.87:5060;branch=z9hG4bK0eaddfef;rport=5060. Record-Route: <sip:10.10.10.154;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. Record-Route: <sip:192.168.1.94;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. From: "Sip User" <sip:11111...@192.168.1.87>;tag=as6b8682ff. To: <sip:22222...@192.168.1.94>;tag=as0736eaf3. Call-ID: 3bb6091225147e4a419fb5ca44e45...@192.168.1.87. CSeq: 102 INVITE. Server: Asterisk Gateway. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Contact: <sip:22222...@10.0.0.54>. Content-Type: application/sdp. Content-Length: 253. . v=0. o=root 1451497710 1451497711 IN IP4 10.0.0.54. s=Asterisk Gateway. c=IN IP4 10.0.0.54. t=0 0. m=audio 19266 RTP/AVP 8 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2010/06/30 09:40:01.655823 192.168.1.94:5060 -> 192.168.1.87:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.87:5060;branch=z9hG4bK0eaddfef;rport=5060. Record-Route: <sip:10.10.10.154;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. Record-Route: <sip:192.168.1.94;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. From: "Sip User" <sip:11111...@192.168.1.87>;tag=as6b8682ff. To: <sip:22222...@192.168.1.94>;tag=as0736eaf3. Call-ID: 3bb6091225147e4a419fb5ca44e45...@192.168.1.87. CSeq: 102 INVITE. Server: Asterisk Gateway. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Contact: <sip:22222...@10.0.0.54>. Content-Type: application/sdp. Content-Length: 273. . v=0. o=root 1451497710 1451497711 IN IP4 192.168.1.94. s=Asterisk Gateway. c=IN IP4 192.168.1.94. t=0 0. m=audio 46066 RTP/AVP 8 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=nortpproxy:yes. U 2010/06/30 09:40:03.650869 10.0.0.54:5060 -> 10.10.10.154:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.10.10.154;branch=z9hG4bK7e45.61f92b7.0;received=10.10.10.154. Via: SIP/2.0/UDP 192.168.1.87:5060;branch=z9hG4bK0eaddfef;rport=5060. Record-Route: <sip:10.10.10.154;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. Record-Route: <sip:192.168.1.94;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. From: "Sip User" <sip:11111...@192.168.1.87>;tag=as6b8682ff. To: <sip:22222...@192.168.1.94>;tag=as0736eaf3. Call-ID: 3bb6091225147e4a419fb5ca44e45...@192.168.1.87. CSeq: 102 INVITE. Server: Asterisk Gateway. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Contact: <sip:22222...@10.0.0.54>. Content-Type: application/sdp. Content-Length: 253. . v=0. o=root 1451497710 1451497711 IN IP4 10.0.0.54. s=Asterisk Gateway. c=IN IP4 10.0.0.54. t=0 0. m=audio 19266 RTP/AVP 8 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2010/06/30 09:40:03.656631 192.168.1.94:5060 -> 192.168.1.87:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.87:5060;branch=z9hG4bK0eaddfef;rport=5060. Record-Route: <sip:10.10.10.154;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. Record-Route: <sip:192.168.1.94;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. From: "Sip User" <sip:11111...@192.168.1.87>;tag=as6b8682ff. To: <sip:22222...@192.168.1.94>;tag=as0736eaf3. Call-ID: 3bb6091225147e4a419fb5ca44e45...@192.168.1.87. CSeq: 102 INVITE. Server: Asterisk Gateway. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Contact: <sip:22222...@10.0.0.54>. Content-Type: application/sdp. Content-Length: 273. . v=0. o=root 1451497710 1451497711 IN IP4 192.168.1.94. s=Asterisk Gateway. c=IN IP4 192.168.1.94. t=0 0. m=audio 46066 RTP/AVP 8 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=nortpproxy:yes. U 2010/06/30 09:40:07.650824 10.0.0.54:5060 -> 10.10.10.154:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.10.10.154;branch=z9hG4bK7e45.61f92b7.0;received=10.10.10.154. Via: SIP/2.0/UDP 192.168.1.87:5060;branch=z9hG4bK0eaddfef;rport=5060. Record-Route: <sip:10.10.10.154;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. Record-Route: <sip:192.168.1.94;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. From: "Sip User" <sip:11111...@192.168.1.87>;tag=as6b8682ff. To: <sip:22222...@192.168.1.94>;tag=as0736eaf3. Call-ID: 3bb6091225147e4a419fb5ca44e45...@192.168.1.87. CSeq: 102 INVITE. Server: Asterisk Gateway. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Contact: <sip:22222...@10.0.0.54>. Content-Type: application/sdp. Content-Length: 253. . v=0. o=root 1451497710 1451497711 IN IP4 10.0.0.54. s=Asterisk Gateway. c=IN IP4 10.0.0.54. t=0 0. m=audio 19266 RTP/AVP 8 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2010/06/30 09:40:07.651026 192.168.1.94:5060 -> 192.168.1.87:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.87:5060;branch=z9hG4bK0eaddfef;rport=5060. Record-Route: <sip:10.10.10.154;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. Record-Route: <sip:192.168.1.94;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. From: "Sip User" <sip:11111...@192.168.1.87>;tag=as6b8682ff. To: <sip:22222...@192.168.1.94>;tag=as0736eaf3. Call-ID: 3bb6091225147e4a419fb5ca44e45...@192.168.1.87. CSeq: 102 INVITE. Server: Asterisk Gateway. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Contact: <sip:22222...@10.0.0.54>. Content-Type: application/sdp. Content-Length: 253. . v=0. o=root 1451497710 1451497711 IN IP4 10.0.0.54. s=Asterisk Gateway. c=IN IP4 10.0.0.54. t=0 0. m=audio 19266 RTP/AVP 8 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2010/06/30 09:40:11.650824 10.0.0.54:5060 -> 10.10.10.154:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.10.10.154;branch=z9hG4bK7e45.61f92b7.0;received=10.10.10.154. Via: SIP/2.0/UDP 192.168.1.87:5060;branch=z9hG4bK0eaddfef;rport=5060. Record-Route: <sip:10.10.10.154;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. Record-Route: <sip:192.168.1.94;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. From: "Sip User" <sip:11111...@192.168.1.87>;tag=as6b8682ff. To: <sip:22222...@192.168.1.94>;tag=as0736eaf3. Call-ID: 3bb6091225147e4a419fb5ca44e45...@192.168.1.87. CSeq: 102 INVITE. Server: Asterisk Gateway. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Contact: <sip:22222...@10.0.0.54>. Content-Type: application/sdp. Content-Length: 253. . v=0. o=root 1451497710 1451497711 IN IP4 10.0.0.54. s=Asterisk Gateway. c=IN IP4 10.0.0.54. t=0 0. m=audio 19266 RTP/AVP 8 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2010/06/30 09:40:11.651001 192.168.1.94:5060 -> 192.168.1.87:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.87:5060;branch=z9hG4bK0eaddfef;rport=5060. Record-Route: <sip:10.10.10.154;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. Record-Route: <sip:192.168.1.94;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. From: "Sip User" <sip:11111...@192.168.1.87>;tag=as6b8682ff. To: <sip:22222...@192.168.1.94>;tag=as0736eaf3. Call-ID: 3bb6091225147e4a419fb5ca44e45...@192.168.1.87. CSeq: 102 INVITE. Server: Asterisk Gateway. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Contact: <sip:22222...@10.0.0.54>. Content-Type: application/sdp. Content-Length: 253. . v=0. o=root 1451497710 1451497711 IN IP4 10.0.0.54. s=Asterisk Gateway. c=IN IP4 10.0.0.54. t=0 0. m=audio 19266 RTP/AVP 8 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2010/06/30 09:40:15.651643 10.0.0.54:5060 -> 10.10.10.154:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.10.10.154;branch=z9hG4bK7e45.61f92b7.0;received=10.10.10.154. Via: SIP/2.0/UDP 192.168.1.87:5060;branch=z9hG4bK0eaddfef;rport=5060. Record-Route: <sip:10.10.10.154;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. Record-Route: <sip:192.168.1.94;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. From: "Sip User" <sip:11111...@192.168.1.87>;tag=as6b8682ff. To: <sip:22222...@192.168.1.94>;tag=as0736eaf3. Call-ID: 3bb6091225147e4a419fb5ca44e45...@192.168.1.87. CSeq: 102 INVITE. Server: Asterisk Gateway. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Contact: <sip:22222...@10.0.0.54>. Content-Type: application/sdp. Content-Length: 253. . v=0. o=root 1451497710 1451497711 IN IP4 10.0.0.54. s=Asterisk Gateway. c=IN IP4 10.0.0.54. t=0 0. m=audio 19266 RTP/AVP 8 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2010/06/30 09:40:15.651863 192.168.1.94:5060 -> 192.168.1.87:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.87:5060;branch=z9hG4bK0eaddfef;rport=5060. Record-Route: <sip:10.10.10.154;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. Record-Route: <sip:192.168.1.94;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. From: "Sip User" <sip:11111...@192.168.1.87>;tag=as6b8682ff. To: <sip:22222...@192.168.1.94>;tag=as0736eaf3. Call-ID: 3bb6091225147e4a419fb5ca44e45...@192.168.1.87. CSeq: 102 INVITE. Server: Asterisk Gateway. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Contact: <sip:22222...@10.0.0.54>. Content-Type: application/sdp. Content-Length: 253. . v=0. o=root 1451497710 1451497711 IN IP4 10.0.0.54. s=Asterisk Gateway. c=IN IP4 10.0.0.54. t=0 0. m=audio 19266 RTP/AVP 8 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2010/06/30 09:40:42.628308 192.168.1.87:5060 -> 192.168.1.94:5060 BYE sip:22222...@10.0.0.54 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.87:5060;branch=z9hG4bK625a7e3b;rport. Route: <sip:192.168.1.94;r2=on;lr=on;ftag=as6b8682ff;nat=yes>,<sip:10.10.10.154;r2=on;lr=on;ftag=as6b8682ff;nat=yes>. From: "Sip User" <sip:11111...@192.168.1.87>;tag=as6b8682ff. To: <sip:22222...@192.168.1.94>;tag=as0736eaf3. Call-ID: 3bb6091225147e4a419fb5ca44e45...@192.168.1.87. CSeq: 103 BYE. User-Agent: Asterisk PBX. Max-Forwards: 70. X-Asterisk-HangupCause: Normal Clearing. X-Asterisk-HangupCauseCode: 16. Content-Length: 0. . U 2010/06/30 09:40:42.630803 10.10.10.154:5060 -> 10.0.0.54:5060 BYE sip:22222...@10.0.0.54 SIP/2.0. Via: SIP/2.0/UDP 10.10.10.154;branch=z9hG4bK8e45.cde34172.0. Via: SIP/2.0/UDP 192.168.1.87:5060;branch=z9hG4bK625a7e3b;rport=5060. From: "Sip User" <sip:11111...@192.168.1.87>;tag=as6b8682ff. To: <sip:22222...@192.168.1.94>;tag=as0736eaf3. Call-ID: 3bb6091225147e4a419fb5ca44e45...@192.168.1.87. CSeq: 103 BYE. User-Agent: Asterisk PBX. Max-Forwards: 69. X-Asterisk-HangupCause: Normal Clearing. X-Asterisk-HangupCauseCode: 16. Content-Length: 0. . U 2010/06/30 09:40:42.638048 10.0.0.54:5060 -> 10.10.10.154:5060 SIP/2.0 481 Call leg/transaction does not exist. Via: SIP/2.0/UDP 10.10.10.154;branch=z9hG4bK8e45.cde34172.0;received=10.10.10.154. Via: SIP/2.0/UDP 192.168.1.87:5060;branch=z9hG4bK625a7e3b;rport=5060. From: "Sip User" <sip:11111...@192.168.1.87>;tag=as6b8682ff. To: <sip:22222...@192.168.1.94>;tag=as0736eaf3. Call-ID: 3bb6091225147e4a419fb5ca44e45...@192.168.1.87. CSeq: 103 BYE. Server: Asterisk Gateway. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Content-Length: 0. . U 2010/06/30 09:40:42.638146 192.168.1.94:5060 -> 192.168.1.87:5060 SIP/2.0 481 Call leg/transaction does not exist. Via: SIP/2.0/UDP 192.168.1.87:5060;branch=z9hG4bK625a7e3b;rport=5060. From: "Sip User" <sip:11111...@192.168.1.87>;tag=as6b8682ff. To: <sip:22222...@192.168.1.94>;tag=as0736eaf3. Call-ID: 3bb6091225147e4a419fb5ca44e45...@192.168.1.87. CSeq: 103 BYE. Server: Asterisk Gateway. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. 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
