[Asterisk-Users] sip disconnects

[snacktime]

I'm trying to figure out if this is a nat problem.

I have a private network behind a freebsd nat box.  The * server is on
a static nat, with a private ip of 10.139.10.165.  I'm connecting with
sjphone as the client from 10.139.10.159.

I am calling out using simpletelecom.  When connecting directly to
simpletelecom using sjphone everything works fine.  When I go through
* I get disconnected after about 20 seconds.  I cannot seem to get my
settings correct, and I don't understand the debug logs enough to know
what's happening.

What I would like to know is what is going on with the following
snippet of the debug log.  Why is * looking for an extension
10.139.10.165?  The only place that string is configured is in the
proxy domain in sjphone.  sip.conf and extensions.conf are at the
bottom.  I can post more debug logs or configs if needed.

Chris

------------------------------------------------------
14 headers, 10 lines
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 8.3.40.113:17398
Found description format GSM
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x2
(gsm)/video=0x0 (nothing), combined - 0x2 (gsm)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined -
0x1 (g723)
set_destination: Parsing
<sip:18006957623 at 63.218.92.199;ftag=as6987e0c2;lr> for address/port to
send to
set_destination: set destination to 63.218.92.199, port 5060
Transmitting:
ACK sip:18006957623 at sip.simpletelecom.com SIP/2.0
Via: SIP/2.0/UDP 10.139.10.165:5060;branch=z9hG4bK30f501e4
Route: <sip:18006957623 at 8.3.40.113:5060>
From: "chris2034" <sip:asterisk at 10.139.10.165>;tag=as6987e0c2
To: <sip:18006957623 at sip.simpletelecom.com>;tag=12E96748-1828
Contact: <sip:2218006957623 at 10.139.10.165>
Call-ID: 06e5f86f2c926b3d394fc3d978176a37 at 10.139.10.165
CSeq: 103 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 63.218.92.199:5060


Sip read: 
OPTIONS sip:10.139.10.165 SIP/2.0
Content-Length: 0
Call-ID: F0620F41-79FA-4917-AE8A-9A3D36589F7C at 10.139.10.159
From: <sip:chris at 10.139.10.165>;tag=1589539025512
CSeq: 25 OPTIONS
Max-Forwards: 70
To: <sip:10.139.10.165>
Via: SIP/2.0/UDP
10.139.10.159;rport;branch=z9hG4bK0a8b0a9f0131c9b1424093c4000078380000008b


8 headers, 0 lines
Looking for 10.139.10.165 in local
Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
10.139.10.159;branch=z9hG4bK0a8b0a9f0131c9b1424093c4000078380000008b
From: <sip:chris at 10.139.10.165>;tag=1589539025512
To: <sip:10.139.10.165>;tag=as1788bc7c
Call-ID: F0620F41-79FA-4917-AE8A-9A3D36589F7C at 10.139.10.159
CSeq: 25 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:10.139.10.165>
Accept: application/sdp
Content-Length: 0


 to 10.139.10.159:5060
Destroying call 'F0620F41-79FA-4917-AE8A-9A3D36589F7C at 10.139.10.159'




sip.conf:
[general]
context=local                   ; Default context for incoming calls
port=5060                       ; UDP Port to bind to (SIP standard
port is 5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls

register => XXXXXXX:XXXXXXX:XXXXXXX at sip.simpletelecom.com/chris2034

[simpleconnect-sip]
type=peer
nat=no
realm=simpletelecom.com
host=sip.simpletelecom.com
username=XXXXXXX
secret=XXXXXXX
dtmfmode=rfc2833

[simpletelecom-incoming]
type=peer
context=local
host=sip.simpletelecom.com


[chris]
nat=yes
context=local
type=friend
host=dynamic
dtmfmode=rfc2833
username=chris
secret=XXXXXXX
canreinvite=no
reinvite=no
callerid="Chris" <6000>
disallow=all
allow=gsm
allow=ulaw


extensions.conf
[simpleconnect]
exten => _22.,1,SetCallerID("XXXXXXX",<Chris>,a)
exten => _22.,2,Dial(SIP/${EXTEN:2}@simpleconnect-sip,30,r)
exten => _22.,3,Hangup()


[local]
include=>simpleconnect
[default]
include = >local