- Remove Key- Removes the account key but leaves the trunks intact.
- Remove Key & Delete Trunks- Removes the account key and removes the trunks.
- Refresh Asterisk Account Info- Updates the module to reflect changes you've made in your SIPStation account. For example, if you purchase a new DID, you will would want to come back to the SIPSTATION module and update the account information by clicking this button followed by the "Apply Config" button.
- Run Firewall Test- Running the firewall test will tell you if you have RTP media port - forwarding set up correctly on your an external firewall. Note that it is recommended that you do set up SIP RTP port forwarding for the full sip media port range, as it will provide more reliable audio for SIP Trunking. You can read more about port forwarding in our wiki "Configuring your PBX or device with SIPStation Service."
- Asterisk Registration Status- Displays the Asterisk Registration status.
- Your Contact IP- This is the contact IP as seen on the gateway. Warnings are provided if errors are detected. This should be your external IP as seen on the WAN side of your router. If it is not, or does not match your Network IP, you should configure your NAT Settings in the Asterisk SIP Settings module or in sip_nat.conf (if not using that module).
- Your Network IP- This is the network IP as seen on the gateway. Warnings are provided if errors are detected. This should be your external IP as seen on the WAN side of your router. If it is not, or it does not match your Network IP, you should configure your NAT Settings in the Asterisk SIP Settings module or in sip_nat.conf (if not using that module.)
- SIP Ping- This is the round-trip signaling delay to the SIP server as determined by the Asterisk ‘qualify’ command. This is signaling delay only. The voice connections (RTP media streams) are routed from your system to the closest POP (Point Of Presence) where the call enters the PSTN. This assures the optimal minimum latency, but can’t be reported because it is dependent on each call.
- Codec Priorities- This is the Codec Priority Asterisk reports for these trunks. This is filtered to show codecs supported by the gateways. The Codecs can be edited on the trunk page to make changes to priority or available codecs.
Firewall Test- Shows the status of your firewall test intended to assist with RTP media port forwarding configuration for external firewall/router devices. When a firewall test is initiated, a free port is chosen from the defined SIP RTP media range on the PBX and an API call is made to the FreePBX mirror server using http/https. The mirror server sends back a sample SIP media payload to the chosen RTP port. If the payload is received successfully, the test passes. This test is not definitive pass/fail. It can assist with config, but may not catch all configuration edge cases for all devices.
The Account Settings area shows you your current settings and allows you to make changes in certain areas. If you make any changes, click the "Update Account Info" button at the bottom of this section, followed by the "Apply Config" button at the top of the module.