Commonly Asked Questions

What Network needs to be open on the firewall and Router?

  • UDP port 5060 and UDP Ports 10,000 - 20,000

 What happens if you have one-way or no-way audio?

  •  Most likely your asterisk server is not sending the proper IP address on the outbound side. Confirm that your router is allowing the proper ports, And confirmed that your asterisk server sending the proper IP. Tools>Asterisk SIP Settings > NAT Settings

What if I have Voice Latency?

  • Check your service to see if it is running out of memory or processing power      
  • if you are using a softphone on your PC, restart device
  • If all is well, check your internet connection          

How to modify CallerID?

  • This is done on either Outbound Routes, Extensions or Trunk. For the Proper string enter: "Name"<number> or just add number

Precautions of changing Voice Codecs?

  •  Be careful, carrier may require different codec types. Using a codec that is not recommended by your provider would cause many of your calls to fail. Once obtaining the required codecs from your provider, you can add them under a few places. You can add them under the trunks. or you can specify then as a global change under Tools>Asterisk SIP Settings.

 Allow server to terminate faxing?

  • Download Fax Configuration module. Turn off "T38 Pass-through"

 Accepting Inbound callerID name?

  • It depends on your carrier, call your carrier to enable this capability. A work around can be enabling CID Superfecta module in FreePBX.