[asterisk-users] Asterisk 1.8 SIP realtime and NAT

[Ishfaq Malik]

Hi

After recently upgrading to 1.8.3 I have noticed that the nat setting
for my peer in my sip table is not making it into the realtime cache.

For example
* Name       : 501
  Realtime peer: Yes, cached
  Secret       : <Set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>
  Context      : pack-local
  Subscr.Cont. : <Not set>
  Language     : 
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 
  Pickupgroup  : 
  MOH Suggest  : 
  Mailbox      : 501 at local
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 5
  Max forwards : 0
  Dynamic      : Yes
  Callerid     : "" <>
  MaxCallBR    : 384 kbps
  Expire       : 3326
  Insecure     : port,invite
  Force rport  : Yes
  ACL          : No
  DirectMedACL : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: -1
  DirectMedia  : Yes
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Text Support : No
  Ign SDP ver  : No
  Trust RPID   : No
  Send RPID    : Yes
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       : 
  Addr->IP     : x.x.x.x:5060
  Defaddr->IP  : (null)
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: PACK501
  SIP Options  : (none)
  Codecs       : 0x10c (ulaw|alaw|g729)
  Codec Order  : (g729:20,alaw:20,ulaw:20)
  Auto-Framing :  No 
  100 on REG   : Yes
  Status       : OK (17 ms)
  Useragent    : snom870/8.4.20
  Reg. Contact : sip:501 at x.x.x.x:52753;line=hu9fedy3
  Qualify Freq : 120000 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess     : 90 secs
  RTP Engine   : asterisk
  Parkinglot   : 
  Use Reason   : No
  Encryption   : No

But in the DB I have clearly set nat to yes

select name,nat from sip where name ='501';
+------+-----+
| name | nat |
+------+-----+
| 501  | yes | 
+------+-----+

In all previous versions of asterisk we have used with realtime we would see a line in the sip show peer looking like: 

Nat          : Always

Has the table definition changed in asterisk 1.8.3?
Is there a bug stopping this value being picked up?

Can someone even point me to the correct source files so I can attempt to try and work out the correct 1.8 sip table definition from there as I can't find one anywhere at all?

Thanks in advance

Ish
-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062