Asterisk 1.8 SIP realtime and NAT

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Asterisk Users 1 Comment

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 :
MD5Secret :
Remote Secret:
Context : pack-local
Subscr.Cont. :
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
MOH Suggest :
Mailbox : 501@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@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

One thought on - Asterisk 1.8 SIP realtime and NAT

  • Hi

    Scratch that

    The value name has changed from Nat to Force Rport

    Back to the drawing board