<-- SIP read from 80.239.235.200:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 200.59.45.204:5060;branch=z9hG4bK06da0607;rport
From: "asterisk" <sip:asterisk@200.59.45.204>;tag=as5bbadca8
To: <sip:sip.voipstunt.com>
Contact: sip:80.239.235.200:5060
Call-ID: 405f7d78607271ab16bcca82407b2bf4@200.59.45.204
CSeq: 102 OPTIONS
Supported:
User-Agent: (Very nice Sip Registrar Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS
Accept: application/sdp
Accept-Encoding:
Accept-Language:

--- (0 headers 1 lines)---
Jul 22 21:40:56 NOTICE[1396]: chan_sip.c:1997 auto_congest: Auto-congesting SIP/voipstunt-0819f668
    -- SIP/voipstunt-0819f668 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'SIP/john-0819a158' status is 'CONGESTION'
Transmitting (NAT) to 201.216.206.221:63635:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.0.4:15540;branch=z9hG4bK-d87543-9b43847bdb2eaf41-1--d87543-;received=201.216.206.221;rport=63635
From: "john"<sip:john@200.59.45.204>;tag=d2427567
To: "+541152184829"<sip:+541152184829@200.59.45.204>;tag=as7968a224
Call-ID: b056c87b6335ce3fOTY5MDU2YWEwNmVmYjMyYjJlNjVjNWFjMmRkOTMyYzk.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:+541152184829@200.59.45.204>
Content-Length: 0


---
asterisk*CLI>
<-- SIP read from 201.216.206.221:63635:
ACK sip:+541152184829@200.59.45.204 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:15540;branch=z9hG4bK-d87543-9b43847bdb2eaf41-1--d87543-;rport
To: "+541152184829"<sip:+541152184829@200.59.45.204>;tag=as7968a224
From: "john"<sip:john@200.59.45.204>;tag=d2427567
Call-ID: b056c87b6335ce3fOTY5MDU2YWEwNmVmYjMyYjJlNjVjNWFjMmRkOTMyYzk.
CSeq: 2 ACK
Content-Length: 0


--- (7 headers 0 lines)---
Destroying call '0b621769696c9a477f5027e105f5018c@200.59.45.204'
Destroying call 'b056c87b6335ce3fOTY5MDU2YWEwNmVmYjMyYjJlNjVjNWFjMmRkOTMyYzk.'