asterisk*CLI> <-- SIP read from 64.34.176.212:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.205:5060;branch=z9hG4bK49d1100c;received=68.149.47.230;rport=5060 From: ;tag=as7d2ab5d0 To: Call-ID: 30e04fa67db52261382a61a6120cb247@192.168.1.205 CSeq: 125 REGISTER User-Agent: LES.NET.VoIP Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- (10 headers 0 lines) --- asterisk*CLI> <-- SIP read from 64.34.176.212:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.205:5060;branch=z9hG4bK49d1100c;received=68.149.47.230;rport=5060 From: ;tag=as7d2ab5d0 To: ;tag=as40f3e2c3 Call-ID: 30e04fa67db52261382a61a6120cb247@192.168.1.205 CSeq: 125 REGISTER User-Agent: LES.NET.VoIP Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 120 Contact: ;expires=120 Date: Mon, 11 Jun 2007 00:00:16 GMT Content-Length: 0 --- (12 headers 0 lines) --- Scheduling destruction of call '30e04fa67db52261382a61a6120cb247@192.168.1.205' in 32000 ms asterisk*CLI> <-- SIP read from 64.34.176.212:5060: INVITE sip:17806283501@68.149.47.230 SIP/2.0 Via: SIP/2.0/UDP 64.34.176.212:5060;branch=z9hG4bK0186bb4d;rport From: "7807097214" ;tag=as52c703ef To: Contact: Call-ID: 0133d3a873b2da4239cc1f8641af4d0c@64.34.176.212 CSeq: 102 INVITE User-Agent: LES.NET.VoIP Max-Forwards: 70 Date: Mon, 11 Jun 2007 00:00:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 216 v=0 o=root 6451 6451 IN IP4 64.34.176.212 s=session c=IN IP4 64.34.176.212 t=0 0 m=audio 10478 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (13 headers 10 lines) --- Using INVITE request as basis request - 0133d3a873b2da4239cc1f8641af4d0c@64.34.176.212 Sending to 64.34.176.212 : 5060 (NAT) Found peer '1213214177' Reliably Transmitting (no NAT) to 64.34.176.212:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 64.34.176.212:5060;branch=z9hG4bK0186bb4d;rport;received=64.34.176.212 From: "7807097214" ;tag=as52c703ef To: ;tag=as68ca19d6 Call-ID: 0133d3a873b2da4239cc1f8641af4d0c@64.34.176.212 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="18850f89" Content-Length: 0 --- Scheduling destruction of call '0133d3a873b2da4239cc1f8641af4d0c@64.34.176.212' in 15000 ms asterisk*CLI> <-- SIP read from 64.34.176.212:5060: ACK sip:17806283501@68.149.47.230 SIP/2.0 Via: SIP/2.0/UDP 64.34.176.212:5060;branch=z9hG4bK0186bb4d;rport From: "7807097214" ;tag=as52c703ef To: ;tag=as68ca19d6 Contact: Call-ID: 0133d3a873b2da4239cc1f8641af4d0c@64.34.176.212 CSeq: 102 ACK User-Agent: LES.NET.VoIP Max-Forwards: 70 Content-Length: 0 --- (10 headers 0 lines) --- asterisk*CLI> <-- SIP read from 64.34.176.212:5060: INVITE sip:17806283501@68.149.47.230 SIP/2.0 Via: SIP/2.0/UDP 64.34.176.212:5060;branch=z9hG4bK40090ebf;rport From: "7807097214" ;tag=as52c703ef To: Contact: Call-ID: 0133d3a873b2da4239cc1f8641af4d0c@64.34.176.212 CSeq: 103 INVITE User-Agent: LES.NET.VoIP Max-Forwards: 70 Proxy-Authorization: Digest username="1213214177", realm="asterisk", algorithm=MD5, uri="sip:17806283501@68.149.47.230", nonce="18850f89", response="9c42c95c021697f7aaeec069ef2e818f", opaque="" Date: Mon, 11 Jun 2007 00:00:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 216 v=0 o=root 6451 6452 IN IP4 64.34.176.212 s=session c=IN IP4 64.34.176.212 t=0 0 m=audio 10478 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (14 headers 10 lines) --- Using INVITE request as basis request - 0133d3a873b2da4239cc1f8641af4d0c@64.34.176.212 Sending to 64.34.176.212 : 5060 (NAT) Found peer '1213214177' Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 64.34.176.212:10478 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 17806283501 in from-sip-external (domain 68.149.47.230) list_route: hop: Transmitting (no NAT) to 64.34.176.212:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.34.176.212:5060;branch=z9hG4bK40090ebf;rport;received=64.34.176.212 From: "7807097214" ;tag=as52c703ef To: Call-ID: 0133d3a873b2da4239cc1f8641af4d0c@64.34.176.212 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing NoOp("SIP/1213214177-0818b020", "Received incoming SIP connection from unknown peer to 17806283501") in new stack -- Executing Set("SIP/1213214177-0818b020", "DID=17806283501") in new stack -- Executing Goto("SIP/1213214177-0818b020", "s|1") in new stack -- Goto (from-sip-external,s,1) -- Executing Ringing("SIP/1213214177-0818b020", "") in new stack Transmitting (no NAT) to 64.34.176.212:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 64.34.176.212:5060;branch=z9hG4bK40090ebf;rport;received=64.34.176.212 From: "7807097214" ;tag=as52c703ef To: ;tag=as2e4fe317 Call-ID: 0133d3a873b2da4239cc1f8641af4d0c@64.34.176.212 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing GotoIf("SIP/1213214177-0818b020", "0?from-trunk|17806283501|1") in new stack -- Executing Set("SIP/1213214177-0818b020", "TIMEOUT(absolute)=15") in new stack -- Channel will hangup at 2007-06-11 00:04:56 UTC. -- Executing Answer("SIP/1213214177-0818b020", "") in new stack We're at 192.168.1.205 port 13810 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 64.34.176.212:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 64.34.176.212:5060;branch=z9hG4bK40090ebf;rport;received=64.34.176.212 From: "7807097214" ;tag=as52c703ef To: ;tag=as2e4fe317 Call-ID: 0133d3a873b2da4239cc1f8641af4d0c@64.34.176.212 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 216 v=0 o=root 3331 3331 IN IP4 192.168.1.205 s=session c=IN IP4 192.168.1.205 t=0 0 m=audio 13810 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Executing Wait("SIP/1213214177-0818b020", "2") in new stack asterisk*CLI> <-- SIP read from 64.34.176.212:5060: ACK sip:17806283501@192.168.1.205 SIP/2.0 Via: SIP/2.0/UDP 64.34.176.212:5060;branch=z9hG4bK341877a6;rport From: "7807097214" ;tag=as52c703ef To: ;tag=as2e4fe317 Contact: Call-ID: 0133d3a873b2da4239cc1f8641af4d0c@64.34.176.212 CSeq: 103 ACK User-Agent: LES.NET.VoIP Max-Forwards: 70 Content-Length: 0 --- (10 headers 0 lines) --- -- Executing Playback("SIP/1213214177-0818b020", "ss-noservice") in new stack -- Playing 'ss-noservice' (language 'en') asterisk*CLI> <-- SIP read from 64.34.176.212:5060: BYE sip:17806283501@192.168.1.205 SIP/2.0 Via: SIP/2.0/UDP 64.34.176.212:5060;branch=z9hG4bK591e2ac2;rport From: "7807097214" ;tag=as52c703ef To: ;tag=as2e4fe317 Call-ID: 0133d3a873b2da4239cc1f8641af4d0c@64.34.176.212 CSeq: 104 BYE User-Agent: LES.NET.VoIP Max-Forwards: 70 Proxy-Authorization: Digest username="1213214177", realm="asterisk", algorithm=MD5, uri="sip:17806283501@192.168.1.205", nonce="18850f89", response="835da58da3de3aa5a733d1dbe3ac7f43", opaque="" Content-Length: 0 --- (10 headers 0 lines) --- Sending to 64.34.176.212 : 5060 (NAT) Transmitting (NAT) to 64.34.176.212:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 64.34.176.212:5060;branch=z9hG4bK591e2ac2;received=64.34.176.212;rport=5060 From: "7807097214" ;tag=as52c703ef To: ;tag=as2e4fe317 Call-ID: 0133d3a873b2da4239cc1f8641af4d0c@64.34.176.212 CSeq: 104 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- == Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/1213214177-0818b020' -- Executing NoOp("SIP/1213214177-0818b020", "Hangup") in new stack -- Executing Set("SIP/1213214177-0818b020", "DID=s") in new stack -- Executing Goto("SIP/1213214177-0818b020", "s|1") in new stack -- Goto (from-sip-external,s,1) -- Executing Ringing("SIP/1213214177-0818b020", "") in new stack -- Executing GotoIf("SIP/1213214177-0818b020", "0?from-trunk|s|1") in new stack -- Executing Set("SIP/1213214177-0818b020", "TIMEOUT(absolute)=15") in new stack -- Channel will hangup at 2007-06-11 00:05:01 UTC. -- Executing Answer("SIP/1213214177-0818b020", "") in new stack == Spawn extension (from-sip-external, s, 4) exited non-zero on 'SIP/1213214177-0818b020' Destroying call '0133d3a873b2da4239cc1f8641af4d0c@64.34.176.212' Destroying call '30e04fa67db52261382a61a6120cb247@192.168.1.205' 12 headers, 0 lines Reliably Transmitting (no NAT) to 192.168.1.198:5060: OPTIONS sip:2000@192.168.1.198:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.205:5060;branch=z9hG4bK19d9c663;rport From: "Unknown" ;tag=as434bc03c To: Contact: Call-ID: 70ff071e37ab3ba349877c5f711bd7e4@192.168.1.205 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 11 Jun 2007 00:05:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- asterisk*CLI> <-- SIP read from 192.168.1.198:5060: SIP/2.0 200 OK To: ;tag=1b057364308db77ei0 From: "Unknown" ;tag=as434bc03c Call-ID: 70ff071e37ab3ba349877c5f711bd7e4@192.168.1.205 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.1.205:5060;branch=z9hG4bK19d9c663;rport=5060 Server: Linksys/PAP2-3.1.12(LS) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura --- (10 headers 0 lines) --- Destroying call '70ff071e37ab3ba349877c5f711bd7e4@192.168.1.205'