Grandstream Setup With AIS VoIP

2 minute read

  1. Gather Supplies
  2. Physical install
    • Place phone.
    • Connect patch cable to patch port in wall to the LAN port on phone.
    • Optional Connect patch cable to computer and the PC port on phone.
    • Troubleshooting: Phone not powering on
      • Ensure the patch port is connected to a PoE on Switch.
      • Check cables are seated and in correct port.
      • If switch is not PoE capable connect a PoE injector on the line or use power cable for the phone itself.
  3. Network connection
    • By default, the phone will use DHCP and should pull IP configurations automatically.
    • Troubleshooting: Phone not getting a network configuration
      • Recheck physical connections.
      • Ensure the phone is not configured with a static IP.
      • Ensure that DHCP server is on the network or DHCP helper addresses have been configured
    • Ensure that device has connectivity with PBX device.
      • Troubleshooting: Phone can not communicate with PBX
      • Check network configurations on network
      • Use Pings, traceroutes, and telnet to narrow down issue.
      • If using the URL of the PBX, verify DNS is functioning properly.
  4. Connect to AIS VoIP
    • Note: these instructions will only implement basic features
    • Applications gt Extensions gt Add Extension gt Add New PJSIP Extension
    • General tab
      • User Extension = An extension you want to put on the phone
      • Display name = can be anything
      • Outbound CID = 10 digit phone number
      • Secret = your discretion but the auto generated one is recommended
    • Voicemail
      • Enable = Yes
      • Voicemail Password = Same as extension
      • If available Email address = User’s Email or Email related to the purpose of the line
  5. Connect to phone
    • Phones setting can be accessed with a browser http://
    • Default login admin/admin you will be prompted to change
    • Document login
    • Accounts gt General Settings
      • Account Name = Name of the line that will appear on the phone
      • SIP server = IP or URL of PBX
      • Outbound Proxy = IP or URL of PBX
      • SIP User ID = Ext in PBX
      • SIP Authentication ID = Ext in PBX
      • Authenticate Password = Copy and Paste extension secret from PBX
      • Name = Whatever you want Caller ID to show
      • Save and Apply
  6. Configure Inbound route
  7. Test

AIS Managed VoIP

As communication methods become more fragmented, office phone use continues to decline but office phone system and service prices have increased in many cases. In designing a robust, reliable, VoIP platform, AIS selected the Asterisk VoIP server software, which is perhaps the most mature and widely deployed VoIP server. Hosting in Amazon Web Services with Geo-Redundant availability zones and connect to the public phone network using geo redundant sip trunks provided by Twilio allows for carrier grade reliability.


Last modified April 16, 2021
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