Cisco settings:
  <natEnabled>true</natEnabled>
  <natAddress>68.93.27.61</natAddress>

Result:
s root@delta:/var/lib/asterisk#/usr/sbin/safe_asterisk: line 138: 22672 Segmentation fault      (core dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} >/dev/${TTY} 2>&1 </dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.

Core filename:
  core.delta.gwp.corp.flightaware.com-2008-11-29T00:43:21-0600

Log and sip debug:

Connected to Asterisk 1.6.0.1 currently running on delta (pid = 22672)
Verbosity was 3 and is now 6
delta*CLI> sip set debug ip 68.93.27.61
SIP Debugging Enabled for IP: 68.93.27.61
delta*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status     
...
nuggetq/nuggetq            (Unspecified)    D   N      5060     UNKNOWN    
peername/peername          (Unspecified)    D   N      5060     UNKNOWN    
...
31 sip peers [Monitored: 17 online, 14 offline Unmonitored: 0 online, 0 offline]
delta*CLI> 
<--- SIP read from TCP://68.93.27.61:34730 --->
REGISTER sip:38.100.147.147 SIP/2.0
Via: SIP/2.0/TCP 68.93.27.61:39048;branch=z9hG4bK8110310a
From: <sip:nuggetq@38.100.147.147>;tag=0021a02bcff40003bcd1d01f-47d17f40
To: <sip:nuggetq@38.100.147.147>
Call-ID: 0021a02b-cff40003-1f502c81-3fbb78a6@68.93.27.61
Max-Forwards: 70
Date: Sat, 29 Nov 2008 06:42:20 GMT
CSeq: 101 REGISTER
User-Agent: Cisco-CP7965G/8.4.0
Contact: <sip:nuggetq@68.93.27.61:39048;transport=tcp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0021a02bcff4>";+u.sip!model.ccm.cisco.com="436"
Supported: replaces,join,norefersub,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-3.0.0,X-cisco-xsi-7.0.1
Content-Length: 0
Expires: 3600


<------------->
--- (13 headers 0 lines) ---
Sending to 68.93.27.61 : 39048 (no NAT)

<--- Transmitting (NAT) to 68.93.27.61:34730 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP 68.93.27.61:39048;branch=z9hG4bK8110310a;received=68.93.27.61
From: <sip:nuggetq@38.100.147.147>;tag=0021a02bcff40003bcd1d01f-47d17f40
To: <sip:nuggetq@38.100.147.147>;tag=as0eb6874a
Call-ID: 0021a02b-cff40003-1f502c81-3fbb78a6@68.93.27.61
CSeq: 101 REGISTER
User-Agent: Asterisk PBX 1.6.0.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="flightaware.com", nonce="265c2c32"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '0021a02b-cff40003-1f502c81-3fbb78a6@68.93.27.61' in 32000 ms (Method: REGISTER)
delta*CLI> 
<--- SIP read from TCP://68.93.27.61:34730 --->
REGISTER sip:38.100.147.147 SIP/2.0
Via: SIP/2.0/TCP 68.93.27.61:39048;branch=z9hG4bKf23b131d
From: <sip:nuggetq@38.100.147.147>;tag=0021a02bcff40003bcd1d01f-47d17f40
To: <sip:nuggetq@38.100.147.147>
Call-ID: 0021a02b-cff40003-1f502c81-3fbb78a6@68.93.27.61
Max-Forwards: 70
Date: Sat, 29 Nov 2008 06:42:20 GMT
CSeq: 102 REGISTER
User-Agent: Cisco-CP7965G/8.4.0
Contact: <sip:nuggetq@68.93.27.61:39048;transport=tcp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0021a02bcff4>";+u.sip!model.ccm.cisco.com="436"
Authorization: Digest username="nuggetq",realm="flightaware.com",uri="sip:38.100.147.147",response="53e55759509c49252581f1a61b66d4b1",nonce="265c2c32",algorithm=MD5
Supported: replaces,join,norefersub,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-3.0.0,X-cisco-xsi-7.0.1
Content-Length: 0
Expires: 3600


<------------->
--- (14 headers 0 lines) ---
Sending to 68.93.27.61 : 34730 (NAT)
Reliably Transmitting (NAT) to 68.93.27.61:34730:
OPTIONS sip:nuggetq@68.93.27.61:39048;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 38.100.147.147:5060;branch=z9hG4bK5c831fd7;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@38.100.147.147>;tag=as150a9859
To: <sip:nuggetq@68.93.27.61:39048;transport=tcp>
Contact: <sip:asterisk@38.100.147.147:5060;transport=TCP>
Call-ID: 1dcf3f8a1f91c6f36e1dbb1b73c1eec6@38.100.147.147
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.1
Date: Sat, 29 Nov 2008 06:42:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


---
    -- Registered SIP 'nuggetq' at 68.93.27.61 port 34730 expires 3600
       > Saved useragent "Cisco-CP7965G/8.4.0" for peer nuggetq

<--- Transmitting (NAT) to 68.93.27.61:34730 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 68.93.27.61:39048;branch=z9hG4bKf23b131d;received=68.93.27.61
From: <sip:nuggetq@38.100.147.147>;tag=0021a02bcff40003bcd1d01f-47d17f40
To: <sip:nuggetq@38.100.147.147>;tag=as0eb6874a
Call-ID: 0021a02b-cff40003-1f502c81-3fbb78a6@68.93.27.61
CSeq: 102 REGISTER
User-Agent: Asterisk PBX 1.6.0.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Expires: 3600
Contact: <sip:nuggetq@68.93.27.61:39048;transport=tcp>;expires=3600
Date: Sat, 29 Nov 2008 06:42:20 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '0021a02b-cff40003-1f502c81-3fbb78a6@68.93.27.61' in 32000 ms (Method: REGISTER)
delta*CLI> 
<--- SIP read from TCP://68.93.27.61:34730 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 38.100.147.147:5060;branch=z9hG4bK5c831fd7;rport
From: "asterisk" <sip:asterisk@38.100.147.147>;tag=as150a9859
To: <sip:nuggetq@68.93.27.61:39048;transport=tcp>;tag=0021a02bcff40004f19c71fe-cf8b857b
Call-ID: 1dcf3f8a1f91c6f36e1dbb1b73c1eec6@38.100.147.147
Date: Sat, 29 Nov 2008 06:42:20 GMT
CSeq: 102 OPTIONS
Server: Cisco-CP7965G/8.4.0
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Allow-Events: kpml,dialog,refer
Accept: application/sdp,multipart/mixed,multipart/alternative
Accept-Encoding: identity
Accept-Language: en
Supported: replaces,join,norefersub,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-3.0.0
Content-Length: 283
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 5839 0 IN IP4 68.93.27.61
s=SIP Call
t=0 0
m=audio 0 RTP/AVP 0 8 18 116 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
--- (17 headers 13 lines) ---
[Nov 29 00:42:20] NOTICE[22766]: chan_sip.c:15231 handle_response_peerpoke: Peer 'nuggetq' is now Reachable. (79ms / 1000ms)
Really destroying SIP dialog '1dcf3f8a1f91c6f36e1dbb1b73c1eec6@38.100.147.147' Method: OPTIONS
delta*CLI> 
<--- SIP read from TCP://68.93.27.61:34730 --->
REGISTER sip:38.100.147.147 SIP/2.0
Via: SIP/2.0/TCP 68.93.27.61:39048;branch=z9hG4bKda82c043
From: <sip:peername@38.100.147.147>;tag=0021a02bcff400029cb8f21c-ea7e40a5
To: <sip:peername@38.100.147.147>
Call-ID: 0021a02b-cff40002-35a1ba72-a8a90e67@68.93.27.61
Max-Forwards: 70
Date: Sat, 29 Nov 2008 06:42:20 GMT
CSeq: 101 REGISTER
User-Agent: Cisco-CP7965G/8.4.0
Contact: <sip:peername@68.93.27.61:39048;transport=tcp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0021a02bcff4>";+u.sip!model.ccm.cisco.com="436"
Supported: replaces,join,norefersub,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-3.0.0,X-cisco-xsi-7.0.1
Content-Length: 0
Reason: SIP;cause=200;text="cisco-alarm:12 Name=SEP0021A02BCFF4 Load=SIP45.8-4-2S Last=cm-reset-tcp"
Expires: 3600


<------------->
--- (14 headers 0 lines) ---
Sending to 68.93.27.61 : 39048 (no NAT)

<--- Transmitting (NAT) to 68.93.27.61:34730 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP 68.93.27.61:39048;branch=z9hG4bKda82c043;received=68.93.27.61
From: <sip:peername@38.100.147.147>;tag=0021a02bcff400029cb8f21c-ea7e40a5
To: <sip:peername@38.100.147.147>;tag=as0a4c4ef5
Call-ID: 0021a02b-cff40002-35a1ba72-a8a90e67@68.93.27.61
CSeq: 101 REGISTER
User-Agent: Asterisk PBX 1.6.0.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="flightaware.com", nonce="7a76b921"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '0021a02b-cff40002-35a1ba72-a8a90e67@68.93.27.61' in 32000 ms (Method: REGISTER)
delta*CLI> 
<--- SIP read from TCP://68.93.27.61:34730 --->
REGISTER sip:38.100.147.147 SIP/2.0
Via: SIP/2.0/TCP 68.93.27.61:39048;branch=z9hG4bK9a659339
From: <sip:peername@38.100.147.147>;tag=0021a02bcff400029cb8f21c-ea7e40a5
To: <sip:peername@38.100.147.147>
Call-ID: 0021a02b-cff40002-35a1ba72-a8a90e67@68.93.27.61
Max-Forwards: 70
Date: Sat, 29 Nov 2008 06:42:21 GMT
CSeq: 102 REGISTER
User-Agent: Cisco-CP7965G/8.4.0
Contact: <sip:peername@68.93.27.61:39048;transport=tcp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0021a02bcff4>";+u.sip!model.ccm.cisco.com="436"
Authorization: Digest username="peername",realm="flightaware.com",uri="sip:38.100.147.147",response="4d5b280b91e94af5d3e16eaafedfb001",nonce="7a76b921",algorithm=MD5
Supported: replaces,join,norefersub,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-3.0.0,X-cisco-xsi-7.0.1
Content-Length: 0
Reason: SIP;cause=200;text="cisco-alarm:12 Name=SEP0021A02BCFF4 Load=SIP45.8-4-2S Last=cm-reset-tcp"
Expires: 3600


<------------->
--- (15 headers 0 lines) ---
Sending to 68.93.27.61 : 34730 (NAT)
Reliably Transmitting (NAT) to 68.93.27.61:34730:
OPTIONS sip:peername@68.93.27.61:39048;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 38.100.147.147:5060;branch=z9hG4bK1bbcf4da;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@38.100.147.147>;tag=as61cf2f5d
To: <sip:peername@68.93.27.61:39048;transport=tcp>
Contact: <sip:asterisk@38.100.147.147:5060;transport=TCP>
Call-ID: 24e3ef165e4ec4d92e5b5398052bb23d@38.100.147.147
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.1
Date: Sat, 29 Nov 2008 06:42:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


---
    -- Registered SIP 'peername' at 68.93.27.61 port 34730 expires 3600
       > Saved useragent "Cisco-CP7965G/8.4.0" for peer peername

<--- Transmitting (NAT) to 68.93.27.61:34730 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 68.93.27.61:39048;branch=z9hG4bK9a659339;received=68.93.27.61
From: <sip:peername@38.100.147.147>;tag=0021a02bcff400029cb8f21c-ea7e40a5
To: <sip:peername@38.100.147.147>;tag=as0a4c4ef5
Call-ID: 0021a02b-cff40002-35a1ba72-a8a90e67@68.93.27.61
CSeq: 102 REGISTER
User-Agent: Asterisk PBX 1.6.0.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Expires: 3600
Contact: <sip:peername@68.93.27.61:39048;transport=tcp>;expires=3600
Date: Sat, 29 Nov 2008 06:42:21 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '0021a02b-cff40002-35a1ba72-a8a90e67@68.93.27.61' in 32000 ms (Method: REGISTER)
delta*CLI> 
<--- SIP read from TCP://68.93.27.61:34730 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 38.100.147.147:5060;branch=z9hG4bK1bbcf4da;rport
From: "asterisk" <sip:asterisk@38.100.147.147>;tag=as61cf2f5d
To: <sip:peername@68.93.27.61:39048;transport=tcp>;tag=0021a02bcff4000509095162-4f8c4f17
Call-ID: 24e3ef165e4ec4d92e5b5398052bb23d@38.100.147.147
Date: Sat, 29 Nov 2008 06:42:21 GMT
CSeq: 102 OPTIONS
Server: Cisco-CP7965G/8.4.0
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Allow-Events: kpml,dialog,refer
Accept: application/sdp,multipart/mixed,multipart/alternative
Accept-Encoding: identity
Accept-Language: en
Supported: replaces,join,norefersub,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-3.0.0
Content-Length: 280
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 5 0 IN IP4 68.93.27.61
s=SIP Call
t=0 0
m=audio 0 RTP/AVP 0 8 18 116 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
5

<------------->
--- (17 headers 14 lines) ---
[Nov 29 00:42:21] NOTICE[22766]: chan_sip.c:15231 handle_response_peerpoke: Peer 'peername' is now Reachable. (79ms / 1000ms)
Really destroying SIP dialog '24e3ef165e4ec4d92e5b5398052bb23d@38.100.147.147' Method: OPTIONS
delta*CLI> 
<--- SIP read from TCP://68.93.27.61:34730 --->
INVITE sip:2500@38.100.147.147 SIP/2.0
Via: SIP/2.0/TCP 68.93.27.61:39048;branch=z9hG4bKeb3072d4
From: "peername" <sip:peername@38.100.147.147>;tag=0021a02bcff400067467ffd3-2b9d1e96
To: <sip:2500@38.100.147.147>
Call-ID: 0021a02b-cff40004-184763dd-41c73160@68.93.27.61
Max-Forwards: 70
Date: Sat, 29 Nov 2008 06:42:45 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7965G/8.4.0
Contact: <sip:peername@68.93.27.61:39048;transport=tcp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Allow-Events: kpml,dialog
Content-Length: 321
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 3322 0 IN IP4 68.93.27.61
s=SIP Call
t=0 0
m=audio 32636 RTP/AVP 0 8 18 116 101
c=IN IP4 68.93.27.61
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<------------->
--- (17 headers 15 lines) ---
  == Using SIP RTP CoS mark 5
Sending to 68.93.27.61 : 39048 (no NAT)
Using INVITE request as basis request - 0021a02b-cff40004-184763dd-41c73160@68.93.27.61
Found user 'peername' for 'peername'

<--- Reliably Transmitting (NAT) to 68.93.27.61:34730 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP 68.93.27.61:39048;branch=z9hG4bKeb3072d4;received=68.93.27.61
From: "peername" <sip:peername@38.100.147.147>;tag=0021a02bcff400067467ffd3-2b9d1e96
To: <sip:2500@38.100.147.147>;tag=as7803183f
Call-ID: 0021a02b-cff40004-184763dd-41c73160@68.93.27.61
CSeq: 101 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="flightaware.com", nonce="27ee51ac"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '0021a02b-cff40004-184763dd-41c73160@68.93.27.61' in 32000 ms (Method: INVITE)
delta*CLI> 
<--- SIP read from TCP://68.93.27.61:34730 --->
ACK sip:2500@38.100.147.147 SIP/2.0
Via: SIP/2.0/TCP 68.93.27.61:39048;branch=z9hG4bKeb3072d4
From: "peername" <sip:peername@38.100.147.147>;tag=0021a02bcff400067467ffd3-2b9d1e96
To: <sip:2500@38.100.147.147>;tag=as7803183f
Call-ID: 0021a02b-cff40004-184763dd-41c73160@68.93.27.61
Date: Sat, 29 Nov 2008 06:42:45 GMT
CSeq: 101 ACK
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---

<--- SIP read from TCP://68.93.27.61:34730 --->
INVITE sip:2500@38.100.147.147 SIP/2.0
Via: SIP/2.0/TCP 68.93.27.61:39048;branch=z9hG4bKd1c9b9b9
From: "peername" <sip:peername@38.100.147.147>;tag=0021a02bcff400067467ffd3-2b9d1e96
To: <sip:2500@38.100.147.147>
Call-ID: 0021a02b-cff40004-184763dd-41c73160@68.93.27.61
Max-Forwards: 70
Date: Sat, 29 Nov 2008 06:42:45 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP7965G/8.4.0
Contact: <sip:peername@68.93.27.61:39048;transport=tcp>
Authorization: Digest username="peername",realm="flightaware.com",uri="sip:2500@38.100.147.147",response="97d056412b21505ff48944578dbd28e7",nonce="27ee51ac",algorithm=MD5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Allow-Events: kpml,dialog
Content-Length: 321
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 3322 0 IN IP4 68.93.27.61
s=SIP Call
t=0 0
m=audio 32636 RTP/AVP 0 8 18 116 101
c=IN IP4 68.93.27.61
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<------------->
--- (18 headers 15 lines) ---
Sending to 68.93.27.61 : 34730 (NAT)
Using INVITE request as basis request - 0021a02b-cff40004-184763dd-41c73160@68.93.27.61
Found user 'peername' for 'peername'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 116
Found RTP audio format 101
Peer audio RTP is at port 68.93.27.61:32636
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format iLBC for ID 116
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x50c (ulaw|alaw|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 68.93.27.61:32636
Looking for 2500 in staff (domain 38.100.147.147)
list_route: hop: <sip:peername@68.93.27.61:39048;transport=tcp>

<--- Transmitting (NAT) to 68.93.27.61:34730 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 68.93.27.61:39048;branch=z9hG4bKd1c9b9b9;received=68.93.27.61
From: "peername" <sip:peername@38.100.147.147>;tag=0021a02bcff400067467ffd3-2b9d1e96
To: <sip:2500@38.100.147.147>
Call-ID: 0021a02b-cff40004-184763dd-41c73160@68.93.27.61
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:2500@38.100.147.147:5060;transport=TCP>
Content-Length: 0


<------------>
    -- Executing [2500@staff:1] NoOp("SIP/peername-08360f60", "Internal Call from Extension "David McNett" <702> ANI 702 for 2500@38.100.147.147") in new stack
    -- Executing [2500@staff:2] Answer("SIP/peername-08360f60", "") in new stack
Audio is at 38.100.147.147 port 10026
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 68.93.27.61:34730 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 68.93.27.61:39048;branch=z9hG4bKd1c9b9b9;received=68.93.27.61
From: "peername" <sip:peername@38.100.147.147>;tag=0021a02bcff400067467ffd3-2b9d1e96
To: <sip:2500@38.100.147.147>;tag=as0b0b6661
Call-ID: 0021a02b-cff40004-184763dd-41c73160@68.93.27.61
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:2500@38.100.147.147:5060;transport=TCP>
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 571317909 571317909 IN IP4 38.100.147.147
s=Asterisk PBX 1.6.0.1
c=IN IP4 38.100.147.147
t=0 0
m=audio 10026 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
    -- Executing [2500@staff:3] VoiceMailMain("SIP/peername-08360f60", "s702") in new stack
    -- <SIP/peername-08360f60> Playing 'vm-youhave.gsm' (language 'en')
delta*CLI> 
<--- SIP read from TCP://68.93.27.61:34730 --->
ACK sip:2500@38.100.147.147:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 68.93.27.61:39048;branch=z9hG4bK3cccb43a
From: "peername" <sip:peername@38.100.147.147>;tag=0021a02bcff400067467ffd3-2b9d1e96
To: <sip:2500@38.100.147.147>;tag=as0b0b6661
Call-ID: 0021a02b-cff40004-184763dd-41c73160@68.93.27.61
Max-Forwards: 70
Date: Sat, 29 Nov 2008 06:42:45 GMT
CSeq: 102 ACK
User-Agent: Cisco-CP7965G/8.4.0
Authorization: Digest username="peername",realm="flightaware.com",uri="sip:2500@38.100.147.147",response="97d056412b21505ff48944578dbd28e7",nonce="27ee51ac",algorithm=MD5
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
delta*CLI> 
<--- SIP read from TCP://68.93.27.61:34730 --->
BYE sip:2500@38.100.147.147:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 68.93.27.61:39048;branch=z9hG4bK3bb2a606
From: "peername" <sip:peername@38.100.147.147>;tag=0021a02bcff400067467ffd3-2b9d1e96
To: <sip:2500@38.100.147.147>;tag=as0b0b6661
Call-ID: 0021a02b-cff40004-184763dd-41c73160@68.93.27.61
Max-Forwards: 70
Date: Sat, 29 Nov 2008 06:42:46 GMT
CSeq: 103 BYE
User-Agent: Cisco-CP7965G/8.4.0
Content-Length: 0
RTP-RxStat: Dur=1,Pkt=36,Oct=6192,LatePkt=0,LostPkt=0,AvgJit=0,VQMetrics="MLQK=0.0000;MLQKav=0.0000;MLQKmn=0.0000;MLQKmx=0.0000;MLQKvr=0.95;CCR=0.0000;ICR=0.0000;ICRmx=0.0000;CS=0;SCS=0"
RTP-TxStat: Dur=1,Pkt=39,Oct=6708
Authorization: Digest username="peername",realm="flightaware.com",uri="sip:2500@38.100.147.147:5060;transport=tcp",response="d91cc26f950fdace381f47bbbb9ea73b",nonce="27ee51ac",algorithm=MD5


<------------->
--- (13 headers 0 lines) ---
Sending to 68.93.27.61 : 34730 (NAT)

<--- Transmitting (NAT) to 68.93.27.61:34730 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 68.93.27.61:39048;branch=z9hG4bK3bb2a606;received=68.93.27.61
From: "peername" <sip:peername@38.100.147.147>;tag=0021a02bcff400067467ffd3-2b9d1e96
To: <sip:2500@38.100.147.147>;tag=as0b0b6661
Call-ID: 0021a02b-cff40004-184763dd-41c73160@68.93.27.61
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.0.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:2500@38.100.147.147:5060;transport=TCP>
Content-Length: 0


<------------>
  == Spawn extension (staff, 2500, 3) exited non-zero on 'SIP/peername-08360f60'
    -- Executing [h@staff:1] NoOp("SIP/peername-08360f60", "Internal Call from Extension "David McNett" <702> ANI 702 for h@38.100.147.147") in new stack
[Nov 29 00:42:46] ERROR[22697]: chan_sip.c:3993 create_addr_from_peer: 'UDP' is not a valid transport for 'peername'. we only use 'TCP'! ending call.
Really destroying SIP dialog '0021a02b-cff40004-184763dd-41c73160@68.93.27.61' Method: BYE
Really destroying SIP dialog '0021a02b-cff40003-1f502c81-3fbb78a6@68.93.27.61' Method: REGISTER
Really destroying SIP dialog '0021a02b-cff40002-35a1ba72-a8a90e67@68.93.27.61' Method: REGISTER
Reliably Transmitting (NAT) to 68.93.27.61:34730:
OPTIONS sip:nuggetq@68.93.27.61:39048;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 38.100.147.147:5060;branch=z9hG4bK72191118;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@38.100.147.147>;tag=as61e64e2a
To: <sip:nuggetq@68.93.27.61:39048;transport=tcp>
Contact: <sip:asterisk@38.100.147.147:5060;transport=TCP>
Call-ID: 734ef59c5ee47d9878f180cb059f8d94@38.100.147.147
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.1
Date: Sat, 29 Nov 2008 06:43:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


---
delta*CLI> 
<--- SIP read from TCP://68.93.27.61:34730 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 38.100.147.147:5060;branch=z9hG4bK72191118;rport
From: "asterisk" <sip:asterisk@38.100.147.147>;tag=as61e64e2a
To: <sip:nuggetq@68.93.27.61:39048;transport=tcp>;tag=0021a02bcff40007b5a07773-a3462618
Call-ID: 734ef59c5ee47d9878f180cb059f8d94@38.100.147.147
Date: Sat, 29 Nov 2008 06:43:20 GMT
CSeq: 102 OPTIONS
Server: Cisco-CP7965G/8.4.0
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Allow-Events: kpml,dialog,refer
Accept: application/sdp,multipart/mixed,multipart/alternative
Accept-Encoding: identity
Accept-Language: en
Supported: replaces,join,norefersub,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-3.0.0
Content-Length: 284
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 22292 0 IN IP4 68.93.27.61
s=SIP Call
t=0 0
m=audio 0 RTP/AVP 0 8 18 116 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
t/8000
a=fmtp:101 0-15
a=sendrecv

<------------->
--- (17 headers 16 lines) ---
delta*CLI> 
Disconnected from Asterisk server
Executing last minute cleanups
s root@delta:/tmp#