Cisco 79xx VoIP Phone Configuration On AIS VoIP

2 minute read

Requirements: Cisco VoIP phones have a maximum Extension Secret/Password length of 16 characters (If a longer password is used, the phone will not register)


  1. create extension for phone in AIS VoIP:
    • Applications gt Extensions gt Add Extension gt New CHAN_SIP
    • User Extension: ext number
    • Display Name: ext number
    • Outbound CID: 10-digit phone number associated
    • Secret:
  2. Click “Advanced”.
    • NAT Mode: Yes
    • Transport: All - TCP Primary
  3. Click “Submit”.
  4. Click “Apply Config” (Red button on top-right).
  5. Add external IP range to firewall port 69:
    • create SEP[MAC ADDRESS].cnf.xml file for phone MAC address.
    • Create phone config file.
    • Open SEPmacaddress.cnf.xml in notepad.
    • Edit gt Replace
    • PASTETEXT with the extension number
    • PASTESECRET with the secret
    • Save file as SEPmacaddress.cnf.xml, replacing macaddress with the device’s actual MAC address with no spaces or dashes.
  6. set phone tftp address to or the associated public IP.
  7. reset phone to factory default if needed.
  8. If desired to have an outside line route directly to the phone, do the following on AIS VoIP Inbound Routing: advanced
    • Signal ringing: yes
    • Force answer: yes
Key Points

  1. Sometimes the factory reset procedure needs to be performed more than once.
  2. 79x0 phones need firmware image name specified in SIPmacaddress.cnf file
  3. All 79xx phones need chan_sip enabled on Extension and configured for port 5160.
  4. All other phone models use chan_pjsip on port 5060.
  5. If having issues with 79x2 phone, check if it has a sticker on the bottom that indicates a “Must use” firmware version. Some indicate “S/W: Must use 9.3(1)SR1 or above”.
Note: You may need to make the following changes on the external pfSense firewall if the phones are unable to provision via TFTP.

  1. System gt Advanced gt Firewall NAT gt TFTP Proxy on LAN interface
  2. Create Manual Outbound Nat Rule. Interface: WAN, Source: Local Subnet, Source Port: UDP, Destination Port 69. Static Port checked.

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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