SIP Debugging enabled
asterisk1*CLI>
<-- SIP read from 192.168.1.77:5060:
REGISTER sip:192.168.1.8 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK1d085bc8ccf5ed22
From: "Bob Roberts" <sip:101@192.168.1.8>;tag=7a491901fbc68e1d
To: <sip:101@192.168.1.8>
Contact: <sip:101@192.168.1.77:5060>
Call-ID: 93057d17f0575345@192.168.1.77
CSeq: 10001 REGISTER
Expires: 3600
User-Agent: Grandstream BT200 1.1.1.14
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0


--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.1.77 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.1.77:5060:
SIP/2.0 404 Not found
Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK1d085bc8ccf5ed22;received=192.168.1.77
From: "Bob Roberts" <sip:101@192.168.1.8>;tag=7a491901fbc68e1d
To: <sip:101@192.168.1.8>;tag=as66da4656
Call-ID: 93057d17f0575345@192.168.1.77
CSeq: 10001 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


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Scheduling destruction of call '93057d17f0575345@192.168.1.77' in 15000 ms
Destroying call '93057d17f0575345@192.168.1.77'