Connect ASTERISK over the WAN

:cry: I have 2 IPPHONES Connected in my LAN . They work perfectly and I can talk to each other with ASTERISK. I took the phone to my house and I configured the phone with my office Static IP. On my office router I have configured the ASTERISK Server with DMZ. Still it gives me NO SERVICE. Can anyone help me?.

Not unless you give us more information, no.

Have you had a read of the relevant pages at:


And Have you searched through old posts on this forum? Because this topic comes up regularly.

This is my sip.conf configuration



This two phones are connected in my office through an ethernet switch with the asterisk server. I can place calls with each other on my LAN. Im also connected to the internet with a LYNKSIS ROUTER BEFS41. I configured the ASTERISK SERVER on DMZ with the ROUTER.
I took one of these phones to my house and I configured the IP Address of the IP Telephone with the STATIC IP that I have in my office. Also at home my internet connection is behind a LYNKSIS ROUTER BEFS41. When I plug the telephone it looks for DHCP and it gives an IP and it’s says NO SERVICE…

If the asterisk server’s in the DMZ, i hope you’ve got some sort of firewall running on the asterisk server! If you have, is this configured to allow connections to the relevant ports?

Can you ping the asterisk server from your computer at home?

Well, that’s no good! You set the IP address of the phone at home to an IP address that’s at your office. The phone will have to have an IP address that’s on your local network - probably something in the subnet if it’s a standard sort of setup.

The phone will also need to be able to connect out through the router. This will depend on how you’ve got the router configured. If the phone can use STUN - and if it’s configured with a useable STUN server - it may be able to negotiate this itself.

What type of phone is it? Have you looked for configuration help with that phone at: … isk+phones

Has anyone compiled stun server for 64bit ? intel or amd

i have
for amd_64 on suse 10.0
no problem

nat=yes in your sip.conf for each of your extensions help any?