gateway*CLI>sip set debug peer 100639

<--- SIP read from 200.203.222.250:60072 --->
NOTIFY sip:189.15.61.128 SIP/2.0
Via: SIP/2.0/UDP 200.203.222.250:60072;branch=z9hG4bK-d41bc38a;rport
From: 100639 <sip:100639@189.15.61.128>;tag=fd942bd905c076ao1
To: <sip:189.15.61.128>
Call-ID: 950dc2a1-11ece8d6@200.203.222.250
CSeq: 164 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-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 event
Via: SIP/2.0/UDP 200.203.222.250:60072;branch=z9hG4bK-d41bc38a;received=200.203.222.250;rport=60072
From: 100639 <sip:100639@189.15.61.128>;tag=fd942bd905c076ao1
To: <sip:189.15.61.128>;tag=as26669cee
Call-ID: 950dc2a1-11ece8d6@200.203.222.250
CSeq: 164 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
Audio is at 189.15.61.128 port 12580
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 200.203.222.250:60072:
INVITE sip:6102@200.203.222.250:60072 SIP/2.0
Via: SIP/2.0/UDP 189.15.61.128:5060;branch=z9hG4bK15a602d3;rport
From: "8132226565" <sip:8132226565@189.15.61.128>;tag=as3350bc6b
To: <sip:6102@200.203.222.250:60072>
Contact: <sip:8132226565@189.15.61.128>
Call-ID: 1a14ce207163f8ab58984ca93bdb253a@189.15.61.128
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 24 Aug 2010 16:49:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 2964 2964 IN IP4 189.15.61.128
s=session
c=IN IP4 189.15.61.128
t=0 0
m=audio 12580 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
<--- SIP read from 200.203.222.250:60072 --->
SIP/2.0 404 Not Found
To: <sip:6102@200.203.222.250:60072>;tag=4f9282d15b5ba666i0
From: "8132226565" <sip:8132226565@189.15.61.128>;tag=as3350bc6b
Call-ID: 1a14ce207163f8ab58984ca93bdb253a@189.15.61.128
CSeq: 102 INVITE
Via: SIP/2.0/UDP 189.15.61.128:5060;branch=z9hG4bK15a602d3;rport=5060
Server: 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/2.0
Via: SIP/2.0/UDP 189.15.61.128:5060;branch=z9hG4bK15a602d3;rport
From: "8132226565" <sip:8132226565@189.15.61.128>;tag=as3350bc6b
To: <sip:6102@200.203.222.250:60072>;tag=4f9282d15b5ba666i0
Contact: <sip:8132226565@189.15.61.128>
Call-ID: 1a14ce207163f8ab58984ca93bdb253a@189.15.61.128
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Really destroying SIP dialog '1a14ce207163f8ab58984ca93bdb253a@189.15.61.128' Method: INVITE