[Asterisk-Users] Asterisk + SIP + NAT - seriously, what's the secret?

[Mark Farver]

Stuart Ford wrote:

>Seriously, this has to be the simplest NAT problem there is with
>Asterisk. What's the secret? How do I learn the dark art? What am I
>missing?
>  
>

I'm guessing here, but the NAT'd grandstream does not have the correct 
external IP configured.

The phones are trying to establish a direct SIP to SIP connection, after 
SIP to SIP call is established asterisk tries to get out of the middle 
of the conversation.  This decreases latency and save processing on the 
asterisk box.  "canreinvite=no" sometimes helps this problem when 
asterisk is a sip client... don't know if it will have an effect here.

The thing to do is setup an extension with the Echo Application.  Call 
that from each phone and see what happens.  If it works for both phones 
you know the problem is a reinvite issue, if one phone or the other 
doesn't work it is a network or Nat config issue.  No sense flailing 
about, try to reduce the problem space.

If your familiar with ethereal it can be used to snoop on the SIP 
connection.. SIP is human readable, so you might be able to learn 
something interesting.

But I really know almost nothing about this.

Mark Farver