I have successfully provisioned several Cisco 7940 phones on my local LAN, and tested functionality. They work properly.
Moving forward, the SIP server was changed in the phones’ configuration files via the endpoint manager, and TFTP server updated in the phone console.
When behind the client router, which has a static public ip, the phone is able to pull the configuration file by TFTP; but SIP cannot register. I see absolutely NOTHING in the asterisk log. The client router is a Linksys WRT54G.
The PBX does not have NAT (static public IP assigned), and does have iptables firewall enabled, with these rules:
# TFTP -A INPUT -m state --state NEW -m udp -p udp --dport tftp -s -j ACCEPT # SIP Port -A INPUT -m state --state NEW -m udp -p udp --dport 5060 -s -j ACCEPT # RTP Ports -A INPUT -m state --state NEW -m udp -p udp --dport 10000:20000 -s -j ACCEPT
In FreePBX/Asterisk SIP Settings, I have:
NAT: no (also tried ‘route’)
IP Configuration: Static IP
External IP: Server public IP
Local Networks: private side network subnets
Reinvite Behavior: Yes
RTP Timers/Ports: default values
Does anybody have any suggestions for what I’ve done incorrectly?