Stuff

polifast2*CLI> 
<-- SIP read from 201.20.205.12:5060: 
INVITE sip:824785501@200.170.105.40 SIP/2.0
Via: SIP/2.0/UDP 201.20.205.12;rport;branch=z9hG4bK41SQScyZF8aUQ
Max-Forwards: 69
From: "Ekiga Teste" <sip:fsrocks@201.20.205.12>;tag=rFQgX2D3HpBcm
To: <sip:824785501@200.170.105.40>
Call-ID: 3b1949e5-9b9f-122a-e2b2-000475e7ee16
CSeq: 85153128 INVITE
Contact: <sip:mod_sofia@201.20.205.12:5060>
User-Agent: FreeSWITCH(mod_sofia)
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
Supported: 100rel, precondition
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 279
Remote-Party-ID: "Ekiga Teste" <sip:fsrocks@201.20.205.12>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 4735461259273311988 7209532826057431998 IN IP4 200.170.105.40
s=FreeSWITCH
c=IN IP4 201.20.205.12
t=0 0
a=sendrecv
m=audio 16388 RTP/AVP 8 8 101 13
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20

--- (15 headers 12 lines) ---
Using INVITE request as basis request - 3b1949e5-9b9f-122a-e2b2-000475e7ee16
Sending to 201.20.205.12 : 5060 (NAT)
Found peer 'fsua'
Found RTP audio format 8
Found RTP audio format 8
Found RTP audio format 101
Found RTP audio format 13
Peer audio RTP is at port 201.20.205.12:16388
Found description format PCMA
Found description format telephone-event
Found description format CN
Capabilities: us - 0x1f07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263|h263p), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event)
Looking for 824785501 in from-sip-extern (domain 200.170.105.40)
Reliably Transmitting (no NAT) to 201.20.205.12:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 201.20.205.12;branch=z9hG4bK41SQScyZF8aUQ;received=201.20.205.12;rport=5060
From: "Ekiga Teste" <sip:fsrocks@201.20.205.12>;tag=rFQgX2D3HpBcm
To: <sip:824785501@200.170.105.40>;tag=as76d4b9db
Call-ID: 3b1949e5-9b9f-122a-e2b2-000475e7ee16
CSeq: 85153128 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
polifast2*CLI> 
<-- SIP read from 201.20.205.12:5060: 
ACK sip:824785501@200.170.105.40 SIP/2.0
Via: SIP/2.0/UDP 201.20.205.12;rport;branch=z9hG4bK41SQScyZF8aUQ
Max-Forwards: 70
From: "Ekiga Teste" <sip:fsrocks@201.20.205.12>;tag=rFQgX2D3HpBcm
To: <sip:824785501@200.170.105.40>;tag=as76d4b9db
Call-ID: 3b1949e5-9b9f-122a-e2b2-000475e7ee16
CSeq: 85153128 ACK
Content-Length: 0


--- (8 headers 0 lines) ---
Destroying call '3b1949e5-9b9f-122a-e2b2-000475e7ee16'
12 headers, 0 lines
Reliably Transmitting (no NAT) to 201.20.205.12:5060:
OPTIONS sip:fsua@201.20.205.12:5060 SIP/2.0
Via: SIP/2.0/UDP 200.170.105.40:5060;branch=z9hG4bK4070ff24;rport
From: "Unknown" <sip:Unknown@200.170.105.40>;tag=as49e5f6b8
To: <sip:fsua@201.20.205.12:5060>
Contact: <sip:Unknown@200.170.105.40>
Call-ID: 355daaea70cd1c1e0ccb8a05139a0b39@200.170.105.40
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 22 Jun 2007 20:08:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
polifast2*CLI> 
<-- SIP read from 201.20.205.12:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 200.170.105.40:5060;branch=z9hG4bK4070ff24;rport=5060
From: "Unknown" <sip:Unknown@200.170.105.40>;tag=as49e5f6b8
To: <sip:fsua@201.20.205.12:5060>;tag=Srg9yXy6eZ1yF
Call-ID: 355daaea70cd1c1e0ccb8a05139a0b39@200.170.105.40
CSeq: 102 OPTIONS
Contact: <sip:201.20.205.12>
User-Agent: FreeSWITCH(mod_sofia)
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
Supported: 100rel, precondition
Content-Length: 0


--- (12 headers 0 lines) ---
Destroying call '355daaea70cd1c1e0ccb8a05139a0b39@200.170.105.40'
polifast2*CLI> sip no debug
SIP Debugging Disabled
polifast2*CLI>