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Linksys PAP2-NA FonoSIP Config

The Linksys Phone Adapter enables use of our high-quality feature-rich telephone service through your cable or DSL Internet connection. Just plug it into your home Router or Gateway and use the two standard telephone jacks to connect your existing phones or fax machines. Each phone jack operates independently, with separate phone service and phone numbers -- like having two phone lines. With, you'll get clear telephone reception and reliable fax connections, even while using the Internet at the same time for normal data operations.

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You must first determine what IP address it received. To do this, you need to pick up the phone attached to the Line 1 jack and dial:

                         **** (four asterisks)

then dial:

                        110 #

and you will be told the IP address of your device (e.g.


Go to any browser equipped computer on your network and enter the address: http://<IP ADDRESS>/
(where <IP ADDRESS> is replaced by the address that was given to you in STEP 1.


Click on the "Admin Login" button near the top right side of the screen, then click on the "Line 1" tab.


You need to modify only a few parameters from the factory default. They are listed here:

Display Name:   Enter your Full name, this will show up as part of your callerid.
User ID:   Enter the phone number you chose when you signed up for service.

Enter the password that you chose when you signed up for service.


To save bandwidth, you can change Line 1 "Preferred Codec" to G729a. You can only do this for one line. So, if Line 1 is on G.729a, Line 2 has to be some other codecs. We do not support G.723.

Audio Configuration


Click on the "Save Settings " button at the bottom of the form.


Make calls!

Behind NAT

If you get one-way audio, you are probably behind NAT. Make the following changes on LINE 1 (you have to click on advanced view to see these options)

Audio Configuration

Behind NAT

On the SIP menu

Audio Configuration


If the phone fails to login, please take the time to double check your configuration as above.
If everything appears to be correct, the problem may be your firewall
  • If your router/firewall suports DMZ, put your hardware phone in the DMZ area
  • If you have an external firewall try opening SIP ports
    SIP signalling ports (UDP) = 5060 - 5061
    DNS port (UDP) = 53
    TFTP port (UDP) = 69
    RTP/RTCP ports (UDP) = 10000 - 30000
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    Powered by iGames supports Xten / Counterpath SIP softphones for Windows, Mac, iPhone and Linux. Internet telephony equipment such as Linksys PAP2, Cisco, Sipura, Nokia N71. Apple iPhone, iPod Touch, iPad. Android. Codec G729. Fring Mobile. Bring your Own Device BYOD. We also support SIP Trunking, replace your phone lines. Asterisk PBX, TrixBox.