[Asterisk-Users] SIP and NAT - best practices?

[Trevor G. Hammonds]

Leo Ann Boon wrote on Saturday, 21 January 2006 6:21 PM:

> Trevor G. Hammonds wrote:
> 
>> How about when you have four or five SIP devices at a single
>> location? Do you manually assign each phone a separate port and add
>> firewall/router rules?  I am looking for an inexpensive device or
>> method that will allow this happen automatically.  Rather than going
>> that route, my current solution is to put an Asterisk server at the
>> client's location to handle the SIP clients and do an outbound
>> trunked IAX connection back to the main server.
>> 
>> 
> Use an outbound proxy either a stanadlone appliance like ix-66 or you
> can build one using Siproxd running on your Linux gateway.
> http://siproxd.sourceforge.net/ 
> 
> There's a WIP port of siproxd to OpenWRT so you can run it on a
> Linksys WRT54G. 

While I have not used siproxd, I have read a bit about it.  From my
understanding of the docs, the local SIP agents register to siproxd, but
siproxd does not register to Asterisk.  So the calls will traverse the NAT
properly, but features like MWI will not work in this scenario.  Also, this
would be pure SIP URL dialling (e.g. usernam at domain.com) as opposed to
traditional telephone dialling (e.g. 1-213-555-8080).  

Please correct me if I am wrong, because I would really like to be (in this
case).  :-)

		Sincerely,
		Trevor Hammonds