Hello, we have for Asterisk servers, and all works fine. But in a few cases we need to register sip phones in one asterisk server from another place, and we can´t register. All the ports are open and redirect to the ¿correct? ports (5060 tcp, and the range we have defined in the phones for udp) in the router, but we can´t register the sip.
Is there any file to configure to solve this problem? Thank you.
hi,
Do you have NAT routers? Are the Asterisk or phone IP’s NAT’d?
If you have NAT at BOTH ends, SIP will not work (without using a STUN server - but not all phones support STUN).
Public IP on the Asterisk server and NAT’d address at handset end should be OK with correct forwarding / configs.
You could overcome any problem by using IAX2 based phone (although there are few to choose from) OR use a site to site VPN between the locations.
We need more info to help further.
Good Luck,
Sean Browne
Cardiff IT Support Ltd
0800 011 2931
www.cardiffitsupport.com
Hello, thank you for your reply. I have registered the sip phone adding the line nat=yes and using a stun server (stun.ekiga.net) but i can hear the other person, but he can´t hear me. What can I do?
Thank you again.
I am using xlite with the probes.
I’d recommend not using Asterisk behind any form of NAT at all if possible.