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Issue: The SIP ID and password aren’t correct.
Resolution: Check the SIP ID and password on the phone.

Issue: There are NAT issues on the firewall or in the Asterisk configuration file.
Resolution: Your personal firewall on your network might have NAT (network address translation) enabled, or it might be blocking the ports that need to be open for VoIP to communicate to the outside world. There are two types of traffic that need to be forwarded: SIP signaling and RTP media. The default port for UDP based SIP signaling is port 5060. The RTP media traffic (the actual audio stream) uses a range of 10000-20000.

Issue: The SIP line isn’t registered.
Resolution: Verify your SIP account registration information in your provisioning letter sent in by IPComms upon sign-up.  Ensure it is the same in your device's SIP account settings.