SIPStation and FAXStation - Ticket Checklist

1) Verify Port Forwarding 

a) Go to Settings → Asterisk SIP Settings and then click on the Chan SIP Settings tab, and then find the option Bind Port. The bind port is the port being used for SIP; typically this will be port 5060. 

b) Staying in Asterisk SIP Settings go back to the General SIP Settings tab, and review the RTP Port Ranges start and stop ports. The typically value will be 10,000 and 20,0000. This is the range of ports that will be used for the media on the call. 

c) Log into your router and verify the SIP port (step a) and RTP Ports (step b) are being forwarded to your SBC. In this example this would mean verifying udp ports 5060 and 10,000-20,000 are forwarded to the PBX IP. You can and should lock down 5060 to be accepted from just our two trunks (trunk1.freepbx.com / 192.159.66.3 and trunk2.freepbx.com / 162.253.134.142)

2) Verify A Inbound Route Is Configured 

If a inbound route isn't setup correctly, you will hear a recording saying there is no service. To verify the routes go to Connectivity → Inbound Routes and ensure a inbound route has been made for the DIDs you have purchased. As well ensure the destinations have been set correctly. The complete guide for inbound routes can be found at Inbound Route User Guide. 

Note, by default routes will be setup to play a DID verification. As a simple test change the destination to a configured extension. Below is an example of this. 

Open