<-- SIP read from 204.147.183.18:5060: 
INVITE sip:104@64.86.22.10 SIP/2.0
Record-Route: <sip:204.147.183.18;ftag=e150392f174cf72febc95185bd47e0a3;lr=on>
Via: SIP/2.0/UDP 204.147.183.18;branch=z9hG4bK0436.57a8e1e5.0
Via: SIP/2.0/UDP 204.147.183.55:1020;branch=z9hG4bKfd79cba87c4de1b175f50da3c4ce06ee;rport=1020
Max-Forwards: 16
From: 33172100200 <sip:33172100200@sip-ny1.stanaphone.com>;tag=e150392f174cf72febc95185bd47e0a3
To: <sip:08152650@sip-ny1.stanaphone.com>
Call-ID: 2d2a20707369f60c24a7d1043656b49d@66.114.240.26
CSeq: 200 INVITE
Contact: Anonymous <sip:204.147.183.55:1020>
Expires: 300
User-Agent: Sippy
cisco-GUID: 3567146758-880725968-1441353657-2263427300
h323-conf-id: 3567146758-880725968-1441353657-2263427300
Content-Length: 286
Content-Type: application/sdp

v=0
o=root 27443 27443 IN IP4 66.114.240.26
s=session
t=0 0
m=audio 56896 RTP/AVP 0 3 18 101
c=IN IP4 204.147.183.10
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=nortpproxy:yes

--- (16 headers 13 lines)---
Using INVITE request as basis request - 2d2a20707369f60c24a7d1043656b49d@66.114.240.26
Sending to 204.147.183.18 : 5060 (non-NAT)
Found no matching peer or user for '204.147.183.18:5060'
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 204.147.183.10:56896
Found description format PCMU
Found description format GSM
Found description format G729
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x106 (gsm|ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 104 in from-sip-external (domain 64.86.22.10)
list_route: hop: <sip:204.147.183.18;ftag=e150392f174cf72febc95185bd47e0a3;lr=on>
Transmitting (no NAT) to 204.147.183.18:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 204.147.183.18;branch=z9hG4bK0436.57a8e1e5.0;received=204.147.183.18
Via: SIP/2.0/UDP 204.147.183.55:1020;branch=z9hG4bKfd79cba87c4de1b175f50da3c4ce06ee;rport=1020
From: 33172100200 <sip:33172100200@sip-ny1.stanaphone.com>;tag=e150392f174cf72febc95185bd47e0a3
To: <sip:08152650@sip-ny1.stanaphone.com>
Call-ID: 2d2a20707369f60c24a7d1043656b49d@66.114.240.26
CSeq: 200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:104@64.86.22.10>
Content-Length: 0


---
    -- Executing NoOp("SIP/sip-ny1.stanaphone.com-4140ffd0", "Received incoming SIP connection from unknown peer to 104") in new stack
    -- Executing Set("SIP/sip-ny1.stanaphone.com-4140ffd0", "DID=104") in new stack
    -- Executing Goto("SIP/sip-ny1.stanaphone.com-4140ffd0", "s|1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing GotoIf("SIP/sip-ny1.stanaphone.com-4140ffd0", "1?from-trunk|104|1") in new stack
    -- Goto (from-trunk,104,1)
Reliably Transmitting (no NAT) to 204.147.183.18:5060:
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 204.147.183.18;branch=z9hG4bK0436.57a8e1e5.0;received=204.147.183.18
Via: SIP/2.0/UDP 204.147.183.55:1020;branch=z9hG4bKfd79cba87c4de1b175f50da3c4ce06ee;rport=1020
From: 33172100200 <sip:33172100200@sip-ny1.stanaphone.com>;tag=e150392f174cf72febc95185bd47e0a3
To: <sip:08152650@sip-ny1.stanaphone.com>;tag=as5a3d1459
Call-ID: 2d2a20707369f60c24a7d1043656b49d@66.114.240.26
CSeq: 200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:104@64.86.22.10>
Content-Length: 0