bee*CLI> sip set debug <--- SIP read from 62.33.2.249:5060 ---> REGISTER sip:195.239.214.91 SIP/2.0 Via: SIP/2.0/UDP 62.33.2.249;branch=z9hG4bK3e2102f90000502246977a0500006a8700005c3a;rport From: "unknown" ;tag=13f1157e2531 To: Contact: Call-ID: 181AF9B64E3D44BA92EE2BC3D57908810x3e2102f9 CSeq: 7876 REGISTER Max-Forwards: 70 User-Agent: SJphone/1.65.377a (SJ Labs) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 62.33.2.249 : 5060 (NAT) <--- Transmitting (no NAT) to 62.33.2.249:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 62.33.2.249;branch=z9hG4bK3e2102f90000502246977a0500006a8700005c3a;received=62.33.2.249;rport=5060 From: "unknown" ;tag=13f1157e2531 To: Call-ID: 181AF9B64E3D44BA92EE2BC3D57908810x3e2102f9 CSeq: 7876 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 62.33.2.249:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 62.33.2.249;branch=z9hG4bK3e2102f90000502246977a0500006a8700005c3a;received=62.33.2.249;rport=5060 From: "unknown" ;tag=13f1157e2531 To: ;tag=as02b62d7b Call-ID: 181AF9B64E3D44BA92EE2BC3D57908810x3e2102f9 CSeq: 7876 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="440bf2f8" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '181AF9B64E3D44BA92EE2BC3D57908810x3e2102f9' in 32000 ms (Method: REGISTER) bee*CLI> sip set debug <--- SIP read from 62.33.2.249:5060 ---> REGISTER sip:195.239.214.91 SIP/2.0 Via: SIP/2.0/UDP 62.33.2.249;branch=z9hG4bK3e2102f90000502346977a050000664500005c3d;rport From: "unknown" ;tag=13f1157e2531 To: Contact: Call-ID: 181AF9B64E3D44BA92EE2BC3D57908810x3e2102f9 CSeq: 7877 REGISTER Max-Forwards: 70 User-Agent: SJphone/1.65.377a (SJ Labs) Content-Length: 0 Authorization: Digest username="214059",realm="asterisk",nonce="440bf2f8",uri="sip:195.239.214.91",response="22db8505983a298f5e7898368eb5ee17",algorithm=MD5 <-------------> --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 62.33.2.249 : 5060 (NAT) <--- Transmitting (no NAT) to 62.33.2.249:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 62.33.2.249;branch=z9hG4bK3e2102f90000502346977a050000664500005c3d;received=62.33.2.249;rport=5060 From: "unknown" ;tag=13f1157e2531 To: Call-ID: 181AF9B64E3D44BA92EE2BC3D57908810x3e2102f9 CSeq: 7877 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 62.33.2.249:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 62.33.2.249;branch=z9hG4bK3e2102f90000502346977a050000664500005c3d;received=62.33.2.249;rport=5060 From: "unknown" ;tag=13f1157e2531 To: ;tag=as02b62d7b Call-ID: 181AF9B64E3D44BA92EE2BC3D57908810x3e2102f9 CSeq: 7877 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="440bf2f8" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '181AF9B64E3D44BA92EE2BC3D57908810x3e2102f9' in 32000 ms (Method: REGISTER) bee*CLI> sip set debug SIP Debugging re-enabled Audio is at 192.168.11.124 port 12378 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.11.125:5060: INVITE sip:214058@192.168.11.125:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK598fc0fb;rport From: "Unknown" ;tag=as5e2afc78 To: Contact: Call-ID: 12dbc60157c26e4710663994234715cd@192.168.11.124 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 13 Jul 2007 13:11:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 13305 13305 IN IP4 192.168.11.124 s=session c=IN IP4 192.168.11.124 t=0 0 m=audio 12378 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- bee*CLI> <--- SIP read from 192.168.11.125:5060 ---> SIP/2.0 100 Trying Via:SIP/2.0/UDP 192.168.11.124:5060;rport;branch=z9hG4bK598fc0fb From:"Unknown" ;tag=as5e2afc78 To: Call-ID:12dbc60157c26e4710663994234715cd@192.168.11.124 CSeq:102 INVITE Content-Type:application/sdp Content-Length:0 <-------------> --- (8 headers 0 lines) --- bee*CLI> <--- SIP read from 192.168.11.125:5060 ---> SIP/2.0 180 Ringing Via:SIP/2.0/UDP 192.168.11.124:5060;rport;branch=z9hG4bK598fc0fb From:"Unknown" ;tag=as5e2afc78 To:;tag=ddafbc27-690448 Call-ID:12dbc60157c26e4710663994234715cd@192.168.11.124 CSeq:102 INVITE Contact: User-Agent:dlink 12-36-7924821 Content-Length:0 <-------------> --- (9 headers 0 lines) --- bee*CLI> <--- SIP read from 192.168.11.125:5060 ---> SIP/2.0 200 OK Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE Via:SIP/2.0/UDP 192.168.11.124:5060;rport;branch=z9hG4bK598fc0fb From:"Unknown" ;tag=as5e2afc78 To:;tag=ddafbc27-690448 Call-ID:12dbc60157c26e4710663994234715cd@192.168.11.124 CSeq:102 INVITE Contact: User-Agent:dlink 12-36-7924821 Content-Type:application/sdp Content-Length:154 v=0 o=214058 1797645180 1797645180 IN IP4 192.168.11.125 s=Session SDP c=IN IP4 192.168.11.125 t=0 0 m=audio 9000 RTP/AVP 8 a=rtpmap:8 PCMA/8000/1 <-------------> --- (11 headers 7 lines) --- Found RTP audio format 8 Peer audio RTP is at port 192.168.11.125:9000 Found description format PCMA for ID 8 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.11.125:9000 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.11.125, port 5060 Transmitting (no NAT) to 192.168.11.125:5060: ACK sip:214058@192.168.11.125:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK33bae570;rport From: "Unknown" ;tag=as5e2afc78 To: ;tag=ddafbc27-690448 Contact: Call-ID: 12dbc60157c26e4710663994234715cd@192.168.11.124 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --e*CLI> [Jul 13 17:11:47] WARNING[15091]: rtp.c:885 ast_rtcp_read: RTCP Read too short bee*CLI> <--- SIP read from 192.168.11.125:5060 ---> BYE sip:Unknown@192.168.11.124 SIP/2.0 Via:SIP/2.0/UDP 192.168.11.125:5060;branch=z9hG4bK8c321c30f4d2abcf From:;tag=ddafbc27-690448 To:"Unknown" ;tag=as5e2afc78 Call-ID:12dbc60157c26e4710663994234715cd@192.168.11.124 CSeq:29 BYE Contact: Max-Forwards:70 User-Agent:dlink 12-36-7924821 Content-Length:0 <-------------> --- (10 headers 0 lines) --- Sending to 192.168.11.125 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.11.125:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.11.125:5060;branch=z9hG4bK8c321c30f4d2abcf;received=192.168.11.125 From: ;tag=ddafbc27-690448 To: "Unknown" ;tag=as5e2afc78 Call-ID: 12dbc60157c26e4710663994234715cd@192.168.11.124 CSeq: 29 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Really destroying SIP dialog '12dbc60157c26e4710663994234715cd@192.168.11.124' Method: BYE Really destroying SIP dialog '181AF9B64E3D44BA92EE2BC3D57908810x3e2102f9' Method: REGISTER bee*CLI> ~~~~~ <--- SIP read from 62.33.2.249:5060 ---> REGISTER sip:195.239.214.91 SIP/2.0 Via: SIP/2.0/UDP 62.33.2.249;branch=z9hG4bK3e2102f90000502a46977a4100006fc900005c40;rport From: "unknown" ;tag=7738157f0f91 To: Contact: Call-ID: 181AF9B64E3D44BA92EE2BC3D57908810x3e2102f9 CSeq: 7878 REGISTER Max-Forwards: 70 User-Agent: SJphone/1.65.377a (SJ Labs) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 62.33.2.249 : 5060 (NAT) <--- Transmitting (no NAT) to 62.33.2.249:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 62.33.2.249;branch=z9hG4bK3e2102f90000502a46977a4100006fc900005c40;received=62.33.2.249;rport=5060 From: "unknown" ;tag=7738157f0f91 To: Call-ID: 181AF9B64E3D44BA92EE2BC3D57908810x3e2102f9 CSeq: 7878 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 62.33.2.249:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 62.33.2.249;branch=z9hG4bK3e2102f90000502a46977a4100006fc900005c40;received=62.33.2.249;rport=5060 From: "unknown" ;tag=7738157f0f91 To: ;tag=as250fc1f7 Call-ID: 181AF9B64E3D44BA92EE2BC3D57908810x3e2102f9 CSeq: 7878 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2351ab89" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '181AF9B64E3D44BA92EE2BC3D57908810x3e2102f9' in 32000 ms (Method: REGISTER) bee*CLI> ~~~~~ <--- SIP read from 62.33.2.249:5060 ---> REGISTER sip:195.239.214.91 SIP/2.0 Via: SIP/2.0/UDP 62.33.2.249;branch=z9hG4bK3e2102f90000502b46977a4100001c0c00005c43;rport From: "unknown" ;tag=7738157f0f91 To: Contact: Call-ID: 181AF9B64E3D44BA92EE2BC3D57908810x3e2102f9 CSeq: 7879 REGISTER Max-Forwards: 70 User-Agent: SJphone/1.65.377a (SJ Labs) Content-Length: 0 Authorization: Digest username="214059",realm="asterisk",nonce="2351ab89",uri="sip:195.239.214.91",response="862930949a3764426e83534214e0d3f8",algorithm=MD5 <-------------> --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 62.33.2.249 : 5060 (NAT) <--- Transmitting (no NAT) to 62.33.2.249:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 62.33.2.249;branch=z9hG4bK3e2102f90000502b46977a4100001c0c00005c43;received=62.33.2.249;rport=5060 From: "unknown" ;tag=7738157f0f91 To: Call-ID: 181AF9B64E3D44BA92EE2BC3D57908810x3e2102f9 CSeq: 7879 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 62.33.2.249:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 62.33.2.249;branch=z9hG4bK3e2102f90000502b46977a4100001c0c00005c43;received=62.33.2.249;rport=5060 From: "unknown" ;tag=7738157f0f91 To: ;tag=as250fc1f7 Call-ID: 181AF9B64E3D44BA92EE2BC3D57908810x3e2102f9 CSeq: 7879 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2351ab89" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '181AF9B64E3D44BA92EE2BC3D57908810x3e2102f9' in 32000 ms (Method: REGISTER) bee*CLI> ~~~~~ <--- SIP read from 192.168.11.123:5060 ---> REGISTER sip:192.168.11.124 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK5F1A0A06 CSeq: 1613 REGISTER To: Expires: 900 From: Call-ID: 1344982143@192.168.11.123 Content-Length: 0 User-Agent: kphone/4.2 Event: registration Allow-Events: presence Contact: ;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER" <-------------> --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.11.123 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.11.123:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK5F1A0A06;received=192.168.11.123 From: To: Call-ID: 1344982143@192.168.11.123 CSeq: 1613 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.11.123:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK5F1A0A06;received=192.168.11.123 From: To: ;tag=as571178af Call-ID: 1344982143@192.168.11.123 CSeq: 1613 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5875e9c6" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1344982143@192.168.11.123' in 32000 ms (Method: REGISTER) bee*CLI> ~~~~~ <--- SIP read from 192.168.11.123:5060 ---> REGISTER sip:192.168.11.124 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK154DDD2 CSeq: 1614 REGISTER To: Authorization: Digest username="214059", realm="asterisk", nonce="5875e9c6", uri="sip:192.168.11.124", cnonce="abcdefghi", nc=00000001, response="89807019c3b6624f07a70fb17c4d4c18", opaque="", algorithm="MD5" Expires: 900 From: Call-ID: 1344982143@192.168.11.123 Content-Length: 0 User-Agent: kphone/4.2 Event: registration Allow-Events: presence Contact: ;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER" <-------------> --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.11.123 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.11.123:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK154DDD2;received=192.168.11.123 From: To: Call-ID: 1344982143@192.168.11.123 CSeq: 1614 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jul 13 17:12:55] WARNING[13334]: db.c:66 dbinit: Unable to open Asterisk database <--- Transmitting (no NAT) to 192.168.11.123:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK154DDD2;received=192.168.11.123 From: To: ;tag=as571178af Call-ID: 1344982143@192.168.11.123 CSeq: 1614 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 900 Contact: ;expires=900 Date: Fri, 13 Jul 2007 13:12:55 GMT Content-Length: 0 <------------>