On Fri, Jul 03, 2009 at 11:45:51AM +0200, Patrick McHardy wrote: > > Joerg Dorchain wrote: >> On Thu, Jul 02, 2009 at 10:17:41AM +0200, Joerg Dorchain wrote: >>> I works, but somewhat ugly. >>> # conntrack -E expect >>> 180 proto=17 src=0.0.0.0 dst=85.93.219.122 sport=0 dport=11080 >>> 180 proto=17 src=0.0.0.0 dst=85.93.219.122 sport=0 dport=11081 >>> 180 proto=17 src=0.0.0.0 dst=212.88.133.153 sport=0 dport=7076 >>> 180 proto=17 src=0.0.0.0 dst=212.88.133.153 sport=0 dport=7077 >>> >>> All the places where the ip is 0.0.0.0 or the port is 0 could be >>> filled more specifically. The necessary information is available >>> in the same SIP/SDP flow as the used information. Besides the two >>> RTP stream are unidirectional, so I'd like to have something like >>> this: >>> 180 proto=17 src=212.88.133.153 dst=85.93.219.122 sport=7076 dport=11080 >>> 180 proto=17 src=85.93.219.122 dst=212.88.133.153 sport=11081 dport=7077 >> >> Sorry for replying to often to myself, I have another addendum: >> In case of asterisk reinvites in order to have to RTP stream >> moved away from the machine, there are still connections expected >> despite that these invites are explicitly meant to stop rtp >> streams to the local machine. > > Could you send a dump? Last time I tried asterisk reinvites, it didn't > work at all because asterisk made some (don't recall the details) > invalid assumptions. Here is the dump. The most interesting lines for you might be those with X-asterisk-Info: SIP re-invite (External RTP bridge) in between. My own asterisk is version 1:1.4.21.2~dfsg-3 from Debian. Bye, Joerg Here is some part missing for setting up the incoming call. The dump starts when asterisk tries the second sip call outgoing, and then directly connect the rtp streams. <-------------> --- (12 headers 0 lines) --- Reliably Transmitting (no NAT) to 217.10.79.9:5060: INVITE sip:10000@xxxxxxxxxx SIP/2.0 Via: SIP/2.0/UDP 212.88.133.153:5060;branch=z9hG4bK7ae9f331;rport From: "Unknown" <sip:1738180@xxxxxxxxxx>;tag=as49b007bc To: <sip:10000@xxxxxxxxxx> Contact: <sip:1738180@xxxxxxxxxxxxxx> Call-ID: 62d694bd2314bbaa7e4ac9470a0a1dfe@xxxxxxxxxx CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 03 Jul 2009 10:55:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 290 v=0 o=root 29312 29312 IN IP4 212.88.133.153 s=session c=IN IP4 212.88.133.153 t=0 0 m=audio 7076 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from 217.10.79.9:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 212.88.133.153:5060;branch=z9hG4bK7ae9f331;rport=5060 From: "Unknown" <sip:1738180@xxxxxxxxxx>;tag=as49b007bc To: <sip:10000@xxxxxxxxxx>;tag=fe1721141f05bd30d4b50c70da3ae228.8296 Call-ID: 62d694bd2314bbaa7e4ac9470a0a1dfe@xxxxxxxxxx CSeq: 102 INVITE Proxy-Authenticate: Digest realm="sipgate.de", nonce="4a4de4c165713d8cf463b184d93833a2d3e9c869" Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (no NAT) to 217.10.79.9:5060: ACK sip:10000@xxxxxxxxxx SIP/2.0 Via: SIP/2.0/UDP 212.88.133.153:5060;branch=z9hG4bK7ae9f331;rport From: "Unknown" <sip:1738180@xxxxxxxxxx>;tag=as49b007bc To: <sip:10000@xxxxxxxxxx>;tag=fe1721141f05bd30d4b50c70da3ae228.8296 Contact: <sip:1738180@xxxxxxxxxxxxxx> Call-ID: 62d694bd2314bbaa7e4ac9470a0a1dfe@xxxxxxxxxx CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Reliably Transmitting (no NAT) to 217.10.79.9:5060: INVITE sip:10000@xxxxxxxxxx SIP/2.0 Via: SIP/2.0/UDP 212.88.133.153:5060;branch=z9hG4bK17f9904c;rport From: "Unknown" <sip:1738180@xxxxxxxxxx>;tag=as49b007bc To: <sip:10000@xxxxxxxxxx> Contact: <sip:1738180@xxxxxxxxxxxxxx> Call-ID: 62d694bd2314bbaa7e4ac9470a0a1dfe@xxxxxxxxxx CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="xxxxxxx", realm="sipgate.de", algorithm=MD5, uri="sip:10000@xxxxxxxxxx", nonce="xxxxxxxxxxxxx", response="xxxxxxxxxxxxxx" Date: Fri, 03 Jul 2009 10:55:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 290 v=0 o=root 29312 29313 IN IP4 212.88.133.153 s=session c=IN IP4 212.88.133.153 t=0 0 m=audio 7076 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from 217.10.79.9:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 212.88.133.153:5060;branch=z9hG4bK17f9904c;rport=5060 From: "Unknown" <sip:1738180@xxxxxxxxxx>;tag=as49b007bc To: <sip:10000@xxxxxxxxxx> Call-ID: 62d694bd2314bbaa7e4ac9470a0a1dfe@xxxxxxxxxx CSeq: 103 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 217.10.79.9:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 212.88.133.153:5060;branch=z9hG4bK17f9904c;rport=5060 Record-Route: <sip:172.20.40.3;lr=on> Record-Route: <sip:217.10.79.9;lr=on;ftag=as49b007bc> From: "Unknown" <sip:1738180@xxxxxxxxxx>;tag=as49b007bc To: <sip:10000@xxxxxxxxxx>;tag=as069f42dc Call-ID: 62d694bd2314bbaa7e4ac9470a0a1dfe@xxxxxxxxxx CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:10000@xxxxxxxxxxxx> Content-Type: application/sdp Content-Length: 287 v=0 o=root 14076 14076 IN IP4 217.10.79.30 s=session c=IN IP4 217.10.79.30 t=0 0 m=audio 16236 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (13 headers 14 lines) --- Transmitting (no NAT) to 217.10.79.9:5060: ACK sip:10000@xxxxxxxxxxxx SIP/2.0 Via: SIP/2.0/UDP 212.88.133.153:5060;branch=z9hG4bK1cf1588f;rport Route: <sip:217.10.79.9;lr=on;ftag=as49b007bc>,<sip:172.20.40.3;lr=on> From: "Unknown" <sip:1738180@xxxxxxxxxx>;tag=as49b007bc To: <sip:10000@xxxxxxxxxx>;tag=as069f42dc Contact: <sip:1738180@xxxxxxxxxxxxxx> Call-ID: 62d694bd2314bbaa7e4ac9470a0a1dfe@xxxxxxxxxx CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Reliably Transmitting (no NAT) to 85.93.219.114:5060: INVITE sip:Unknown@xxxxxxxxxxxxx SIP/2.0 Via: SIP/2.0/UDP 212.88.133.153:5060;branch=z9hG4bK24a401b3;rport Route: <sip:85.93.219.114;ftag=as271217c9;lr=on> From: <sip:20400371@xxxxxx>;tag=as67860b84 To: "Unknown" <sip:Unknown@xxxxxxxxxxxxx>;tag=as271217c9 Contact: <sip:s@xxxxxxxxxxxxxx> Call-ID: 130aaf675f1c3e327fc3860c35001e11@xxxxxxxxxxxxx CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 287 v=0 o=root 29312 29313 IN IP4 217.10.79.30 s=session c=IN IP4 217.10.79.30 t=0 0 m=audio 16236 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 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 --- Reliably Transmitting (no NAT) to 217.10.79.9:5060: INVITE sip:10000@xxxxxxxxxxxx SIP/2.0 Via: SIP/2.0/UDP 212.88.133.153:5060;branch=z9hG4bK04dd2986;rport Route: <sip:217.10.79.9;lr=on;ftag=as49b007bc>,<sip:172.20.40.3;lr=on> From: "Unknown" <sip:1738180@xxxxxxxxxx>;tag=as49b007bc To: <sip:10000@xxxxxxxxxx>;tag=as069f42dc Contact: <sip:1738180@xxxxxxxxxxxxxx> Call-ID: 62d694bd2314bbaa7e4ac9470a0a1dfe@xxxxxxxxxx CSeq: 104 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 289 v=0 o=root 29312 29314 IN IP4 85.93.219.122 s=session c=IN IP4 85.93.219.122 t=0 0 m=audio 13006 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from 85.93.219.114:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 212.88.133.153:5060;branch=z9hG4bK24a401b3;rport=5060 From: <sip:20400371@xxxxxx>;tag=as67860b84 To: "Unknown" <sip:Unknown@xxxxxxxxxxxxx>;tag=as271217c9 Call-ID: 130aaf675f1c3e327fc3860c35001e11@xxxxxxxxxxxxx CSeq: 102 INVITE Server: Sip EXpress router (0.9.6 (x86_64/linux)) Content-Length: 0 Warning: 392 85.93.219.114:5060 "Noisy feedback tells: pid=31420 req_src_ip=212.88.133.153 req_src_port=5060 in_uri=sip:Unknown@xxxxxxxxxxxxx out_uri=sip:Unknown@xxxxxxxxxxxxx via_cnt==1" <-------------> --- (9 headers 0 lines) --- <--- SIP read from 85.93.219.114:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 212.88.133.153:5060;branch=z9hG4bK24a401b3;rport=5060 Record-Route: <sip:85.93.219.114;ftag=as67860b84;lr=on> From: <sip:20400371@xxxxxx>;tag=as67860b84 To: "Unknown" <sip:Unknown@xxxxxxxxxxxxx>;tag=as271217c9 Call-ID: 130aaf675f1c3e327fc3860c35001e11@xxxxxxxxxxxxx CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:Unknown@xxxxxxxxxxxxx> Content-Type: application/sdp Content-Length: 312 v=0 o=root 1491947058 1491947059 IN IP4 85.93.219.122 s=Asterisk PBX 1.6.0.6 c=IN IP4 85.93.219.122 t=0 0 m=audio 13006 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (13 headers 14 lines) --- Transmitting (no NAT) to 85.93.219.114:5060: ACK sip:Unknown@xxxxxxxxxxxxx SIP/2.0 Via: SIP/2.0/UDP 212.88.133.153:5060;branch=z9hG4bK1f7a1716;rport Route: <sip:85.93.219.114;ftag=as271217c9;lr=on> From: <sip:20400371@xxxxxx>;tag=as67860b84 To: "Unknown" <sip:Unknown@xxxxxxxxxxxxx>;tag=as271217c9 Contact: <sip:s@xxxxxxxxxxxxxx> Call-ID: 130aaf675f1c3e327fc3860c35001e11@xxxxxxxxxxxxx CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- <--- SIP read from 217.10.79.9:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 212.88.133.153:5060;branch=z9hG4bK04dd2986;rport=5060 From: "Unknown" <sip:1738180@xxxxxxxxxx>;tag=as49b007bc To: <sip:10000@xxxxxxxxxx>;tag=as069f42dc Call-ID: 62d694bd2314bbaa7e4ac9470a0a1dfe@xxxxxxxxxx CSeq: 104 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 217.10.79.9:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 212.88.133.153:5060;branch=z9hG4bK04dd2986;rport=5060 From: "Unknown" <sip:1738180@xxxxxxxxxx>;tag=as49b007bc To: <sip:10000@xxxxxxxxxx>;tag=as069f42dc Call-ID: 62d694bd2314bbaa7e4ac9470a0a1dfe@xxxxxxxxxx CSeq: 104 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:10000@xxxxxxxxxxxx> Content-Type: application/sdp Content-Length: 287 v=0 o=root 14076 14077 IN IP4 217.10.79.30 s=session c=IN IP4 217.10.79.30 t=0 0 m=audio 16236 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (11 headers 14 lines) --- Transmitting (no NAT) to 217.10.79.9:5060: ACK sip:10000@xxxxxxxxxxxx SIP/2.0 Via: SIP/2.0/UDP 212.88.133.153:5060;branch=z9hG4bK3e4efa0c;rport Route: <sip:217.10.79.9;lr=on;ftag=as49b007bc>,<sip:172.20.40.3;lr=on> From: "Unknown" <sip:1738180@xxxxxxxxxx>;tag=as49b007bc To: <sip:10000@xxxxxxxxxx>;tag=as069f42dc Contact: <sip:1738180@xxxxxxxxxxxxxx> Call-ID: 62d694bd2314bbaa7e4ac9470a0a1dfe@xxxxxxxxxx CSeq: 104 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- <--- SIP read from 217.10.79.9:5060 ---> BYE sip:1738180@xxxxxxxxxxxxxx SIP/2.0 Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bKf4e9.3b548123.0 Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bKf4e9.3b548123.0 Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK15d17460;rport=5060 From: <sip:10000@xxxxxxxxxx>;tag=as069f42dc To: "Unknown" <sip:1738180@xxxxxxxxxx>;tag=as49b007bc Call-ID: 62d694bd2314bbaa7e4ac9470a0a1dfe@xxxxxxxxxx CSeq: 102 BYE Max-Forwards: 68 Content-Length: 0 X-hint: rr-enforced <-------------> --- (11 headers 0 lines) --- Sending to 217.10.79.9 : 5060 (no NAT) <--- Transmitting (no NAT) to 217.10.79.9:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bKf4e9.3b548123.0;received=217.10.79.9 Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bKf4e9.3b548123.0 Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK15d17460;rport=5060 From: <sip:10000@xxxxxxxxxx>;tag=as069f42dc To: "Unknown" <sip:1738180@xxxxxxxxxx>;tag=as49b007bc Call-ID: 62d694bd2314bbaa7e4ac9470a0a1dfe@xxxxxxxxxx CSeq: 102 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:1738180@xxxxxxxxxxxxxx> Content-Length: 0 <------------> Reliably Transmitting (no NAT) to 85.93.219.114:5060: INVITE sip:Unknown@xxxxxxxxxxxxx SIP/2.0 Via: SIP/2.0/UDP 212.88.133.153:5060;branch=z9hG4bK273424fd;rport Route: <sip:85.93.219.114;ftag=as271217c9;lr=on> From: <sip:20400371@xxxxxx>;tag=as67860b84 To: "Unknown" <sip:Unknown@xxxxxxxxxxxxx>;tag=as271217c9 Contact: <sip:s@xxxxxxxxxxxxxx> Call-ID: 130aaf675f1c3e327fc3860c35001e11@xxxxxxxxxxxxx CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 290 v=0 o=root 29312 29314 IN IP4 212.88.133.153 s=session c=IN IP4 212.88.133.153 t=0 0 m=audio 7074 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 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 --- <--- SIP read from 85.93.219.114:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 212.88.133.153:5060;branch=z9hG4bK273424fd;rport=5060 From: <sip:20400371@xxxxxx>;tag=as67860b84 To: "Unknown" <sip:Unknown@xxxxxxxxxxxxx>;tag=as271217c9 Call-ID: 130aaf675f1c3e327fc3860c35001e11@xxxxxxxxxxxxx CSeq: 103 INVITE Server: Sip EXpress router (0.9.6 (x86_64/linux)) Content-Length: 0 Warning: 392 85.93.219.114:5060 "Noisy feedback tells: pid=31389 req_src_ip=212.88.133.153 req_src_port=5060 in_uri=sip:Unknown@xxxxxxxxxxxxx out_uri=sip:Unknown@xxxxxxxxxxxxx via_cnt==1" <-------------> --- (9 headers 0 lines) --- <--- SIP read from 85.93.219.114:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 212.88.133.153:5060;branch=z9hG4bK273424fd;rport=5060 Record-Route: <sip:85.93.219.114;ftag=as67860b84;lr=on> From: <sip:20400371@xxxxxx>;tag=as67860b84 To: "Unknown" <sip:Unknown@xxxxxxxxxxxxx>;tag=as271217c9 Call-ID: 130aaf675f1c3e327fc3860c35001e11@xxxxxxxxxxxxx CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.0.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:Unknown@xxxxxxxxxxxxx> Content-Type: application/sdp Content-Length: 312 v=0 o=root 1491947058 1491947060 IN IP4 85.93.219.122 s=Asterisk PBX 1.6.0.6 c=IN IP4 85.93.219.122 t=0 0 m=audio 13006 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (13 headers 14 lines) --- Transmitting (no NAT) to 85.93.219.114:5060: ACK sip:Unknown@xxxxxxxxxxxxx SIP/2.0 Via: SIP/2.0/UDP 212.88.133.153:5060;branch=z9hG4bK688e4031;rport Route: <sip:85.93.219.114;ftag=as271217c9;lr=on> From: <sip:20400371@xxxxxx>;tag=as67860b84 To: "Unknown" <sip:Unknown@xxxxxxxxxxxxx>;tag=as271217c9 Contact: <sip:s@xxxxxxxxxxxxxx> Call-ID: 130aaf675f1c3e327fc3860c35001e11@xxxxxxxxxxxxx CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Reliably Transmitting (no NAT) to 85.93.219.114:5060: BYE sip:Unknown@xxxxxxxxxxxxx SIP/2.0 Via: SIP/2.0/UDP 212.88.133.153:5060;branch=z9hG4bK352dc0dd;rport Route: <sip:85.93.219.114;ftag=as271217c9;lr=on> From: <sip:20400371@xxxxxx>;tag=as67860b84 To: "Unknown" <sip:Unknown@xxxxxxxxxxxxx>;tag=as271217c9 Call-ID: 130aaf675f1c3e327fc3860c35001e11@xxxxxxxxxxxxx CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- <--- SIP read from 85.93.219.114:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 212.88.133.153:5060;branch=z9hG4bK352dc0dd;rport=5060 Record-Route: <sip:85.93.219.114;ftag=as67860b84;lr=on> From: <sip:20400371@xxxxxx>;tag=as67860b84 To: "Unknown" <sip:Unknown@xxxxxxxxxxxxx>;tag=as271217c9 Call-ID: 130aaf675f1c3e327fc3860c35001e11@xxxxxxxxxxxxx CSeq: 104 BYE User-Agent: Asterisk PBX 1.6.0.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 <-------------> --- (11 headers 0 lines) ---
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