[Asterisk-Users] Asterisk behind NAT -- SIP config file

[Anton Krall]

>I am using Polycom SP300 phones. You have to separate 'user' and 'peer'
part of it to >>>get it working. Search the wiki for description of the
problem.

Nice to know ... I don't own any of those but its good general knowledge.

>You have to forward port 5060 so that phone from outside can register and
call. And >>>>>ports 10000-20000 do that voice can go through. Actual port
ranfge is isn filr >>>>>>rtp.conf.> 10000-20000 is  the default range

Ive done this on the firewall infront of our * box. 

>Yes, only port 5060. If you do not forward 5060, you can not call this
phone 
>from outside. Seem to work OK without other ports being forwarded.

You mean on the remote sip phone firewall? What if there arem ore than 1 sip
phone on that network behidn that firewall?

Don't you need to forward ports 10000-20000 for voice? Or does the sip
phones just open up the ports from inside (by doing the in to out calls and
keep alives)?