<-- SIP read from 10.10.10.16:5060:
INVITE sip:106@ SIP/2.0
Via: SIP/2.0/UDP 10.10.10.16;branch=z9hG4bK2F7815AF
CSeq: 1390 INVITE
To: <sip:106@>
Content-Type: application/sdp
From: "Asterisk" <sip:@>;tag=6AF9841B
Call-ID: 436630706@10.10.10.16
Subject: sip:@
Content-Length: 183
User-Agent: kphone/4.1.0
Contact: "Asterisk" <sip:@10.10.10.16;transport=udp>

v=0
o=username 0 0 IN IP4 10.10.10.16
s=The Funky Flow
c=IN IP4 10.10.10.16
t=0 0
m=audio 58354 RTP/AVP 97 3 0
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000

--- (11 headers 9 lines) ---
Ignoring this INVITE request
We're at 10.10.10.104 port 12516
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x400 (ilbc) to SDP
Reliably Transmitting (no NAT) to 10.10.10.16:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.16;branch=z9hG4bK2F7815AF;received=10.10.10.16
From: "Asterisk" <sip:@>;tag=6AF9841B
To: <sip:106@>;tag=as35b3d76a
Call-ID: 436630706@10.10.10.16
CSeq: 1390 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:106@10.10.10.104>
Content-Type: application/sdp
Content-Length: 207

v=0
o=root 2490 2496 IN IP4 10.10.10.104
s=session
c=IN IP4 10.10.10.104
t=0 0
m=audio 12516 RTP/AVP 0 3 97
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=silenceSupp:off - - - -