Q: All my calls are getting disconnected after 30 seconds, one minute or five minutes
A: During a call our system verifys connection periodically. When this verification fails, it is normally because your device is behind a NAT router or firewall and the NAT traversal is not enabled.
If you have Grandstream IP Phone (Budgetone) or Grandstream ATA286 (Handytone) check on the web configuration page that you have "NAT Traversal" set to "Yes" (leave "Stun Server" blank), "Random Port" set to "No" and "Keep-Alive interval" to "7". Please also ensure that "TFTP Server" is set to "63.214.186.4".
With a Cisco ATA186 check also on the device web page.
Make sure you have these settings:
NatServer: calamar0.nikotel.com
NatTimer: 0x0000007
Outbound Proxy: calamar0.nikotel.com
GkOrgateway: calamar0.nikotel.com
Both the Grandstream IP Phone and Cisco ATA186 need to be restarted to have the changes activated. Unplug the power and reconnect it after three minutes.
Both nikotel4mac/nikotel4win do not need any special configuration and do not have this problem in common. If you experience these issue with on of them, please check that "I'm using DSL/NAT Router" and "Use Keep Alive" is checked at the "Setup/Network" Pane (at the Preferences with Nikotel4Mac) and set to "7" seconds or less.
For Microsoft messenger there is no work around available. The Microsoft messenger does not work from behind a NAT router. This is a limitation of the Microsoft messenger and not of the nikotel network. Other softphones like nikotel4win work from behind a NAT router.
For SJphone you need to configure port forward or port mapping.
If all fail then do you need a port forwarding or better a strict port mapping on your router.Port mapping example (this example configuration may not match your required network settings):
For Grandstream (all products)
settings for phone 1:
static IP: 192.168.0.100
local SIP port: 5000
local RTP port 5001, use random port: no, nat: yes (leave stun blank!)
settings for phone 2:
static IP: 192.168.0.101
local SIP port: 5003
local RTP port 5004, use random port: no, nat: yes (leave stun blank!)
For Cisco (ATA186):
DHCP: 0
StaticIP: 192.168.0.100
StaticRoute: 192.168.0.1 (this is the IP of the NAT router)
StaticNetMask: 255.255.255.0
SipPort: 5000
Mediaport: 5001
For NIkotel4Win/Nikotel4Mac:
The main settings for these products are read from the network adapter.
You have to configure your Windows or Mac System to "IP" "192.168.0.100", 'Route(r)" to "192.168.0.1", "NetMask" to "255.255.255.0".
Then start nikotel4Win or Nikotel4Mac and go to the "Setup/Network" pane and then set "Local SIP port" to "5000" and "Server SIP port" to 5060.
Please switch to the "Voice" pane then and set "RTP port low" to "5001" and "RTP port high" to "5001" also. Please press "save" now.Router (IP 192.168.0.1):
map the ports UDP 5000 and UDP 5001 to IP 192.168.0.100
map the ports UDP 5003 and UDP 5004 to IP 192.168.0.100If you need support for another product then please open a support ticket at Nikotel.
Q: I have a grandstream phone and my calls get interrupted after
aprox. 60 seconds
A: Clear the "Outbound proxy" field at the web configuration interface
of the phone. Update and then reboot the phone.
