[asterisk-users] Asterisk and NAT ?

[yusuf]

Noc Phibee wrote:
> 
> Hi
> 
> I am search a small information
> 
> - i use Asterisk on official IP without Nat
> 
> - My first VoIP phone are a Thomson 2030 on a NAT Network.
>   That's work very good.
> 
> 
> But now, i want add a second phone, a Linksys SPA-941 on
> the same network of the Thomson 2030 ...
> 
> My problems that i don't see a solution into asterisk or
> on my firewall for that's work.
> 
> When i call to the thomson, that's work, when i call to the linksys
> that's don't ring ...
> 
> On my asterisk i have put :
> 200= thomson
> 202= linksys
> 
> 
> [200]
> port=5060
> username=200
> secret=X
> type=friend
> host=dynamic
> disallow=all
> allow=g729
> allow=alaw
> allow=ulaw
> context=interne
> qualify=yes
> nat=route
> dtmfmode=rfc2833
> language=fr
> 
> 
> [202]
> port=5070
> username=202
> secret=X
> type=friend
> host=dynamic
> disallow=all
> allow=g729
> allow=alaw
> allow=ulaw
> context=interne
> nat=route
> dtmfmode=rfc2833
> language=fr
> 
> 
> 
> on my firewall, i have put a forward of port 5060 to thomson and 5070 to 
> linksys
> in UDP and TCP.
> 
> On linksys i can call but not receive call
> on thomson i can call and receive without problems
> 

Hi,

you dont have to/should'nt be using different SIP ports for each phone.  Its completely not needed. 
  Also, you dont have/need to port forward.  Just open ports 5060 and 1000-20000, on the box that 
asterisk is running, and on your NAT router. Dont port forward.

Then in sip.conf


  [202]
  username=202
  secret=X
  type=friend
  host=dynamic
  disallow=all
  allow=g729
  allow=alaw
  allow=ulaw
  context=interne
  nat=yes
  canreinvite=no	


  [200]
  username=200
  secret=X
  type=friend
  host=dynamic
  disallow=all
  allow=g729
  allow=alaw
  allow=ulaw
  context=interne
  nat=yes
  canreinvite=no	


then restart linksys and thomson, and you will see that they both register on asterisk cli.  Now you 
will be able to call/receive on both.

-- 
thanks,
yusuf

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