Sip read: INVITE sip:301@204.213.176.174;user=phone SIP/2.0 Via: SIP/2.0/UDP 204.213.176.211:5060;branch=z9hG4bK7853ba4b From: "Anonymous" ;tag=003094c384f7002a343a9067-72d4bd77 To: Call-ID: 003094c3-84f70024-39d4e3d6-196e4538@204.213.176.211 Date: Mon, 08 Nov 2004 00:01:30 GMT CSeq: 101 INVITE User-Agent: CSCO/7 Contact: Expires: 180 Content-Type: application/sdp Content-Length: 253 Accept: application/sdp v=0 o=Cisco-SIPUA 20332 11783 IN IP4 204.213.176.211 s=SIP Call c=IN IP4 204.213.176.211 t=0 0 m=audio 31680 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 13 headers, 11 lines Using latest request as basis request Sending to 204.213.176.211 : 5060 (NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 204.213.176.211:31680 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x10e(GSM|ULAW|ALAW|G729A), peer - audio=0x10c(ULAW|ALAW|G729A)/video=0x0(EMPTY), combined - 0x10c(ULAW|ALAW|G729A) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found peer '311f' Reliably Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 204.213.176.211:5060;branch=z9hG4bK7853ba4b;received=204.213.176.211;rport=5060 From: "Anonymous" ;tag=003094c384f7002a343a9067-72d4bd77 To: ;tag=as11f569f3 Call-ID: 003094c3-84f70024-39d4e3d6-196e4538@204.213.176.211 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="6075e071" Content-Length: 0 to 204.213.176.211:5060 Scheduling destruction of call '003094c3-84f70024-39d4e3d6-196e4538@204.213.176.211' in 15000 ms asterisk*CLI> Sip read: ACK sip:301@204.213.176.174;user=phone SIP/2.0 Via: SIP/2.0/UDP 204.213.176.211:5060;branch=z9hG4bK7853ba4b From: "Anonymous" ;tag=003094c384f7002a343a9067-72d4bd77 To: ;tag=as11f569f3 Call-ID: 003094c3-84f70024-39d4e3d6-196e4538@204.213.176.211 Date: Mon, 08 Nov 2004 00:01:31 GMT CSeq: 101 ACK Content-Length: 0 8 headers, 0 lines asterisk*CLI> Sip read: INVITE sip:301@204.213.176.174;user=phone SIP/2.0 Via: SIP/2.0/UDP 204.213.176.211:5060;branch=z9hG4bK6b965403 From: "Anonymous" ;tag=003094c384f7002a343a9067-72d4bd77 To: Call-ID: 003094c3-84f70024-39d4e3d6-196e4538@204.213.176.211 Date: Mon, 08 Nov 2004 00:01:31 GMT CSeq: 102 INVITE User-Agent: CSCO/7 Contact: Proxy-Authorization: Digest username="311f",realm="asterisk",uri="sip:204.213.176.174",response="1fd45cc71af3dca2ed841fa3b6a3dc96",nonce="6075e071",algorithm=md5 Expires: 180 Content-Type: application/sdp Content-Length: 253 v=0 o=Cisco-SIPUA 20332 11783 IN IP4 204.213.176.211 s=SIP Call c=IN IP4 204.213.176.211 t=0 0 m=audio 31680 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 13 headers, 11 lines Using latest request as basis request Sending to 204.213.176.211 : 5060 (NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 204.213.176.211:31680 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x10e(GSM|ULAW|ALAW|G729A), peer - audio=0x10c(ULAW|ALAW|G729A)/video=0x0(EMPTY), combined - 0x10c(ULAW|ALAW|G729A) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found peer '311f' asterisk*CLI> Disconnected from Asterisk server /usr/sbin/safe_asterisk: line 83: 16606 Killed asterisk ${CLIARGS} ${ASTARGS} 1>&/dev/${TTY}