2 asterisk servers (remote), 2 sip phones (remote), 1 DID

Just trying to see if this is possible, definitely willing to do the research on my own to make it work, just asking if it can be done, although, if it can, a little inside from some experts would be much appreciated. Have searched but cant seem to find the question that I’m asking.

A little background first:
I have a server at my house running A@H. I get two IP address’ from my ISP. One IP is given to a Linksys WRT54GS v1.1 that contains all the PC in my house that run Windows. The other IP is normally given to a box running ipcop, which has a green interface that dishes out ip’s to all of my linux pc’s behind it via dhcp. With A@H behind the ipcop box, I can forward ports to the box, and make and recieve calls fine, using a softphone from any other system being ipcop, i just purchased a GXP-2000 to be configured on the ipcop network, and used with Asterisk. If I forgo ipcop and connect the A@H server directly to the internet so that it recieves a public IP, and connect my Window’s PC directly to the internet (no router) I can connect to it from remote locations and it works fine as well. So its obvious that NAT is my issue.

I would like to have my friend, who uses the same ISP to be able to connect and use my A@H box as well, using a GXP-2000 as well. I am aware of the serious issues with NAT and SIP, and have attempted to work them out, but so far to no avail, but my question isnt about that. My question is simply, rather than trying to get NAT Traversal working out, could we instead, put up an A@H (or real Asterisk, doesnt matter) at his location (behind a Linksys WRT54G Router) and use IAX2 to have the two A@H computer’s communicate, but still use the SIP phones and have one DID on the A@H computer on my side? But still be able to dial each others extensions and have it ring as if we were connected to the same A@H server? And when he dial’s out it uses my DID, and dial’s through my A@H server? So basically the only thing his A@H box would be used for would be to help with the SIP/NAT problem.

Any insight into this, or any other suggestion would be much appreciated, thanks!

It is not that hard …

Did u get a dynamic public ip? if so consider subscribing to a dynamic DNS provider near you so that www.whatever.com should point to your adsl router.

Then have the router forward/dmz that incoming request from internet to your a@h server ip address 192.x.x.x.

your friend can then just point his sip phone to www.whatever.com

I recieve static public IP’s so registering a dynamic DNS would be pointless. The problem doesnt lie with the A@H boxes, but rather the Linksys WRT54G, as even when the A@H box is behind ipcop, with SIP and RTP ports forwarded, if a remote computer that is directly exposed to the internet (ie we remove his WRT54G and plug his machine directly into his modem) so that there is no NAT or firewall issue in between, he can connect fine, but rather when he is behind the WRT54G, it seems that no matter what ports are forwarded, or firewall off, or dmz, he cannot connect to my A@H box.

So my question still remains, could we use two A@H box’s, say set one up as a user and the other as a peer, share dial plans, and a DID number, and when I dial his extension on his A@H box, it would ring his SIP phone. So have the two servers communicate with IAX, but have our phones and DID number and use SIP termination and origination.

you can connect the 2 Asterisk boxes and do what ever routing you want between them (such as what you described). However - you should be able to get the remote SIP phone to connect to your Asterisk box and it will probably be less work than setting up the 2 Asterisk boxes and all your desired routing.

Did you make sure you had all the appropriate ports forwarded back to your box, and the approrpiate settings in your sip.conf file (like nat=yes, etc.).

Also - make sure your Linksys router has the lates firmware. I read a posting from someone who was having headaches getting his Telasip service running (SIP) and a firmware upgrade ultimately fixed the problem. He had the same router as you.



I can confirm the firmware problem.

My father bought a WRT54G V5 with firmware 1.00.1 and when he installed
it, his grandstream stopped being able to log into my asterisk server via SIP.

Upgrading the firmware to 1.00.9 solved the problem.


Hi all,

I too had firmware problems in the past. I also heard about a dynamic dns service for $5.75 a year you get up to one domain and five host names at dns.t4tm.net/usign.asp. They offer a trial period for up to 7 days too, add your domain and download a client and that’s it, the client keeps your ip updated automatically.

Hope this helps someone as I have received a lot of help since my short time.

[quote=“tlofton1000”]Hi all,

Link works better after removing the period from the end: dns.t4tm.net/usign.asp

Hope this helps someone as I have received a lot of help since my short time.[/quote]