Something

proxy*CLI>
    -- Executing [502@home:1] Dial("SIP/501-08e46698", "SIP/502") in new stack
Audio is at 192.168.101.1 port 16714
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.101.97:5060:
INVITE sip:502@192.168.101.97 SIP/2.0
Via: SIP/2.0/UDP 192.168.101.1:5060;branch=z9hG4bK783ee380;rport
From: "Ryans" <sip:501@192.168.101.1>;tag=as5453cc8f
To: <sip:502@192.168.101.97>
Contact: <sip:501@192.168.101.1>
Call-ID: 7b11b51166590bde387e2b96773e1c34@192.168.101.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 18 Apr 2007 09:10:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 4508 4508 IN IP4 192.168.101.1
s=session
c=IN IP4 192.168.101.1
t=0 0
m=audio 16714 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 502
proxy*CLI>
<--- SIP read from 192.168.101.97:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.101.1:5060;branch=z9hG4bK783ee380;rport
To: <sip:502@192.168.101.97>
From: "Ryans" <sip:501@192.168.101.1>;tag=as5453cc8f
Call-ID: 7b11b51166590bde387e2b96773e1c34@192.168.101.1
CSeq: 102 INVITE
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
proxy*CLI>
<--- SIP read from 192.168.101.97:5060 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 192.168.101.1:5060;branch=z9hG4bK783ee380;rport
To: <sip:502@192.168.101.97>;tag=6f2mld3f2hhc7icbd0qk
From: "Ryans" <sip:501@192.168.101.1>;tag=as5453cc8f
Call-ID: 7b11b51166590bde387e2b96773e1c34@192.168.101.1
CSeq: 102 INVITE
Warning: 304 192.168.101.97 Media type not available
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 192.168.101.97:5060:
ACK sip:502@192.168.101.97 SIP/2.0
Via: SIP/2.0/UDP 192.168.101.1:5060;branch=z9hG4bK783ee380;rport
From: "Ryans" <sip:501@192.168.101.1>;tag=as5453cc8f
To: <sip:502@192.168.101.97>;tag=6f2mld3f2hhc7icbd0qk
Contact: <sip:501@192.168.101.1>
Call-ID: 7b11b51166590bde387e2b96773e1c34@192.168.101.1
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
    -- SIP/502-08e48b60 is circuit-busy
Scheduling destruction of SIP dialog '7b11b51166590bde387e2b96773e1c34@192.168.101.1' in 39296 ms (Method: INVITE)
Reliably Transmitting (NAT) to 192.168.101.97:5060:
BYE sip:502@192.168.101.97 SIP/2.0
Via: SIP/2.0/UDP 192.168.101.1:5060;branch=z9hG4bK21efd426;rport
From: "Ryans" <sip:501@192.168.101.1>;tag=as5453cc8f
To: <sip:502@192.168.101.97>;tag=6f2mld3f2hhc7icbd0qk
Call-ID: 7b11b51166590bde387e2b96773e1c34@192.168.101.1
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'SIP/501-08e46698' status is 'CONGESTION'
proxy*CLI>
<--- SIP read from 192.168.101.97:5060 --->
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 192.168.101.1:5060;branch=z9hG4bK21efd426;rport
To: <sip:502@192.168.101.97>;tag=6f2mld3f2hhc7icbd0qk
From: "Ryans" <sip:501@192.168.101.1>;tag=as5453cc8f
Call-ID: 7b11b51166590bde387e2b96773e1c34@192.168.101.1
CSeq: 103 BYE
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
[Apr 18 19:10:41] WARNING[4538]: chan_sip.c:12311 handle_response: Remote host can't match request BYE to call '7b11b51166590bde387e2b96773e1c34@192.168.101.1'. Giving up.
Really destroying SIP dialog '7b11b51166590bde387e2b96773e1c34@192.168.101.1' Method: INVITE
proxy*CLI> exit
[root@proxy asterisk]#