[tphank]

Any problems having a Phonegnome and SPA3000 on the same network sharing a single internet connection?

How about 2 SPA3000?

[mberlant]

If you have sufficient bandwidth and a cooperative router (not Symmetric NAT) there should be no conflict between any number of SIP devices behind the router. Your router will play traffic cop and keep all of the registrations and conversations clear of each other.

I presently have one GS286, one SPA-2000, one SPA3000, one SPA-841 and an Asterisk server all behind the same WRT54GS router with no complications.

[yiannis]

It also helps to setup the ATAs, to use different ports than 5060 when they register with several SIP proxies.

Increase the port by 2 as in 5060, 5062, 5064 etc.

[mberlant]

Why would you need to bother changing the listening port on either device when they have different LAN IP addresses? Your router can quite easily distinguish between 192.168.0.100:5060 and 192.168.0.101:5060.

[tphank]

I have both the 3000 and PhoneGnome working great side by side. I now want to move my AAH machine home. Any problems with this, considering that I will be setting up the router to forward the necessary ports for the AAH for use by remote systems/extensions: SIP 5004-5082; RTP 10001-20000; IAX 4569?

I have my and other phonegnomes successfully using this AAH system( for best rate routing) , but it doesn't do it locally, so the AAH system has to have the ports forwarded.
Thanks

[mberlant]

I have my Asterisk behind my NAT router inside my home and have experienced no problems occasionally registering client SIP devices to "other" servers through the same router.

I am confused, though, by your choice of SIP ports in your list. Since Asterisk only listens on UDP 5060 for inbound client (extension) SIP Registrations, why are you forwarding 5004-5059 and 5061-5082 as well?

[tphank]

I was just following the suggestions in the NerdVittles.com step by step... if 5060 alone is enough, I will try it.
What about the RTP?
THanks

[mberlant]

I did not notice that advice on nerdvittles, but I cannot see any reason to forward any port through the router that will never receive an unsolicited desired packet. As far as I know, 5060 (unless you change it) is the only SIP control port Asterisk listens to, so it's the only one that should be forwarded through the router.

rtp.conf says 10000-20000, so that is what I use. If you don't expect to have 5000 simultaneous conversations you can reduce this range to what you like. I have not done so on the assumption that there are devices out in the marketplace that prefer to use different bits of this range and may not behave properly elsewhere.