sip debug

gateway*CLI>sip set debug peer 100639<--- SIP read from 200.203.222.250:60072 --->NOTIFY sip:189.15.61.128 SIP/2.0Via: SIP/2.0/UDP 200.203.222.250:60072;branch=z9hG4bK-d41bc38a;rportFrom: 100639 <sip:100639@189.15.61.128>;tag=fd942bd905c076ao1To: <sip:189.15.61.128>Call-ID: 950dc2a1-11ece8d6@200.203.222.250CSeq: 164 NOTIFYMax-Forwards: 70Event: keep-aliveUser-Agent: Linksys/PAP2T-3.1.15(LS)Content-Length: 0<------------->--- (10 headers 0 lines) ---Sending to 200.203.222.250 : 60072 (NAT)<--- Transmitting (NAT) to 200.203.222.250:60072 --->SIP/2.0 489 Bad eventVia: SIP/2.0/UDP 200.203.222.250:60072;branch=z9hG4bK-d41bc38a;received=200.203.222.250;rport=60072From: 100639 <sip:100639@189.15.61.128>;tag=fd942bd905c076ao1To: <sip:189.15.61.128>;tag=as26669ceeCall-ID: 950dc2a1-11ece8d6@200.203.222.250CSeq: 164 NOTIFYUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Length: 0<------------>Audio is at 189.15.61.128 port 12580Adding codec 0x100 (g729) to SDPAdding non-codec 0x1 (telephone-event) to SDPReliably Transmitting (NAT) to 200.203.222.250:60072:INVITE sip:6102@200.203.222.250:60072 SIP/2.0Via: SIP/2.0/UDP 189.15.61.128:5060;branch=z9hG4bK15a602d3;rportFrom: "8132226565" <sip:8132226565@189.15.61.128>;tag=as3350bc6bTo: <sip:6102@200.203.222.250:60072>Contact: <sip:8132226565@189.15.61.128>Call-ID: 1a14ce207163f8ab58984ca93bdb253a@189.15.61.128CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Tue, 24 Aug 2010 16:49:14 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 2964 2964 IN IP4 189.15.61.128s=sessionc=IN IP4 189.15.61.128t=0 0m=audio 12580 RTP/AVP 18 101a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv---<--- SIP read from 200.203.222.250:60072 --->SIP/2.0 404 Not FoundTo: <sip:6102@200.203.222.250:60072>;tag=4f9282d15b5ba666i0From: "8132226565" <sip:8132226565@189.15.61.128>;tag=as3350bc6bCall-ID: 1a14ce207163f8ab58984ca93bdb253a@189.15.61.128CSeq: 102 INVITEVia: SIP/2.0/UDP 189.15.61.128:5060;branch=z9hG4bK15a602d3;rport=5060Server: Linksys/PAP2T-3.1.15(LS)Content-Length: 0<------------->--- (8 headers 0 lines) ---Transmitting (NAT) to 200.203.222.250:60072:ACK sip:6102@200.203.222.250:60072 SIP/2.0Via: SIP/2.0/UDP 189.15.61.128:5060;branch=z9hG4bK15a602d3;rportFrom: "8132226565" <sip:8132226565@189.15.61.128>;tag=as3350bc6bTo: <sip:6102@200.203.222.250:60072>;tag=4f9282d15b5ba666i0Contact: <sip:8132226565@189.15.61.128>Call-ID: 1a14ce207163f8ab58984ca93bdb253a@189.15.61.128CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0---Really destroying SIP dialog '1a14ce207163f8ab58984ca93bdb253a@189.15.61.128' Method: INVITE