[Asterisk-Users] realtime sip show peers no nat

[G.Marshall]

Hello

sip show peers does not mark hosts as NAT even though sip.conf and
sip_peers table has nat=yes.

spitfire*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Mask            
Port     Status
voipuser.org/gdsm          216.127.66.119       N      255.255.255.255 
5060     Unmonitored
5560/5560                  192.168.4.5      D   N   A  255.255.255.255 
5060     Unmonitored
5561/5561                  192.168.4.5      D   N   A  255.255.255.255 
5061     Unmonitored
4561/4561                  212.74.112.53    D   N      255.255.255.255 
8413     Unmonitored
4 sip peers [4 online , 0 offline]
spitfire*CLI>

asterisk listens on 192.168.4.3 and 82.70.154.145.  The host 212.74.112.53
is the external (NAT) address for a sip phone whose LAN address is
10.44.16.163.

sip debug shows the following
spitfire*CLI>
<-- SIP read from 212.74.112.53:8413:
REGISTER sip:82.70.154.145 SIP/2.0
Via: SIP/2.0/UDP 10.44.16.163:5060
From: <sip:4561 at 82.70.154.145;user=phone>;tag=2361964166
To: <sip:4561 at 82.70.154.145;user=phone>
Call-ID: 1812954233 at 10.44.16.163
CSeq: 1 REGISTER
Contact: <sip:4561 at 10.44.16.163:5060;user=phone;transport=udp>;expires=120
User-Agent: Cisco ATA 186  v3.1.0 atasip (040211A)
Content-Length: 0


--- (9 headers 0 lines)---
Using latest request as basis request
Sending to 10.44.16.163 : 5060 (NAT)
Transmitting (NAT) to 212.74.112.53:8413:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.44.16.163:5060;received=212.74.112.53;rport=8413
From: <sip:4561 at 82.70.154.145;user=phone>;tag=2361964166
To: <sip:4561 at 82.70.154.145;user=phone>
Call-ID: 1812954233 at 10.44.16.163
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 120
Contact: <sip:4561 at 82.70.154.145>;expires=120
Content-Length: 0


---
Transmitting (NAT) to 212.74.112.53:8413:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.44.16.163:5060;received=212.74.112.53;rport=8413
From: <sip:4561 at 82.70.154.145;user=phone>;tag=2361964166
To: <sip:4561 at 82.70.154.145;user=phone>;tag=as0771f231
Call-ID: 1812954233 at 10.44.16.163
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 120
Contact: <sip:4561 at 10.44.16.163:5060;user=phone;transport=udp>;expires=120
Date: Fri, 13 May 2005 01:59:09 GMT
Content-Length: 0


---
Scheduling destruction of call '1812954233 at 10.44.16.163' in 15000 ms


Does anyone know how to rectify this?  By the looks of things,

Many thanks,

Spencer