[Asterisk-Users] sip invite question

[Matt Hess]

I receive an invite from a vendor device..

U 2005/09/19 09:00:18.991139 66.185.96.32:5060 -> 66.185.96.23:5060
   INVITE sip:3034580009 at 66.185.96.23;userphone SIP/2.0..Via: 
SIP/2.0/UDP voip.livewirenet.com:5060;branch=z9hG4bK6ee64f86
Max-Forwards: 70
To: "3036284320" <sip:3036284320 at voip.livewirenet.com>;tag=as60bbddbc
From: <sip:3034580009 at 66.185.96.32>;user=phone;tag=286349056
Call-ID: 78074cc132c4ac664728aba75ac82854 at voip.livewirenet.com
CSeq: 104 INVITE
Contact: sip:66.185.96.32:5060
Content-Type: application/sdp
Content-Length: 101

v=0
o=- 1127142018 1127142018 IN IP4 0.0.0.0
s=-
c=IN IP4 0.0.0.0
t=0 0
m=audio 4010 RTP/AVP 0


Then I see with (sip debug on) that asterisk sends an ok to itself.. why 
does it do this? My speculation is that the above invite is doing 
something incorrect.. but what?



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