[Asterisk-Users] Re: qualify and NAT....

[Eric Wieling aka ManxPower]

>>Remember clients send packets from a random high port number which 
>>changes.  Port forwarding on your router is pretty useless.  nat=yes 
>>combined with qualify=yes should cause enough traffic on the right 
>>ports to keep the NAT translations open on your NAT router.

Brian McCrary wrote:
> 
> In theory, it sounds as if should work pretty easily, since there is a
> connection already established between the ATA and Asterisk.  One big
> think I forgot to mention is my NAT device is a MC3810 with a PRI
> attached to it, so it already is running it's own SIP user agent.  So,
> when the ATA gets a call, Asterisk forwards it to the MC3810 (since it's
> IP registered with the ATA through NAT.)  Since the MC3810 doesn't know
> about the ATAs it just returns "404 Not Found", and that's the end of
> it.
> 
> So, I'm guessing, perhaps having a SIP user agent running on the NAT
> router itself could be causing some of the problem?  It's frustrating to
> see Asterisk talking to the phone with OPTIONS packets, but then not be
> able to send a call to it, seems like I must be close.

Everything I said assumes that the NAT router is NOT "SIP Aware".  On 
the Ciscos I manage I turn off their "SIP fixup" feature.  Mostly 
because I want them to act like Linksys, or whatever, that our users 
have at home.

If your NAT router is "SIP Aware", then you don't need portforwarding, 
qualify=yes or nat=yes.  You should have nat=no because the NAT router 
is doing all the magic and Asterisk's NAT magic may not work with the 
router's NAT magic.

Search the archives for "MC3810" or other keywords relaying to the 
router.  I seem to recall someone else mentioning the same model.

Search Asterisk mailing lists by prepending site:lists.digium.com to 
your Google search.  Browse the mailing list archive at 
http://lists.digium.com/


-- 
Eric Wieling * BTEL Consulting * 504-210-3699 x2120