VoIP
Search| Help | My Account
Sign Up Now
  • HOME
  • HOW IT WORKS
  • FEATURES
  • PRICING
  • CALLING RATES
  • PHONE CARD
  • SUPPORT

Home / Support / Configuration Guides / Linksys PAP2

Linksys PAP2 VoIP settings & configuration GuideLinksys PAP2

The Linksys PAP2 VoIP 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. Each phone jack operates independently, with separate phone service and phone numbers - like having two phone lines. With VoIPVoIP, you'll get clear telephone reception, even while using the Internet at the same time for normal data operations

STEP 1

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. 192.168.0.100)

STEP 2

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.

STEP 3

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

linksys pap2 configuration step 3

 

STEP 4

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

Proxy: sip3.voipvoip.com

Display Name: Enter your full name, this will show up as part of your caller ID.

User ID: Enter the account number assigned to you when signed up for VoIPVoIP service.

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

linksys pap2 configuration 4

 

STEP 5

To save bandwidth, you can change Line 1 "Preferred Codec" to G729a. Also change the "Use Pref Codec Only" to No. 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.

linksys pap2 configuration step 5

 

STEP 6

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

linksys configuration step 6

 

STEP 7

Make Calls!

Problems? If you get one-way audio, or cannot get a dial tone or cannot make/receive calls with your VoIP device, you are probably behind firewall. Your router's firewall (also known as NAT) is blocking certain operations of the VoIP telephone adapter or some of the settings of the adapter is not entered correctly.

If you are sure that your account number & password is entered correctly to your device, before trying to resolve this issue on your router or voip device, you can get all device settings (other than account number and password) remotely from our servers.

In order to receive configurations remotely go to VOICE/ ADVANCED and find PROVISIONING tab and change the field below:

- Profile Rule: http://config.voipvoip.com/$PN

Click on the "Submit All Changes" button at the bottom of the form. Unplug your device from the power and plug back after 5 seconds to reboot the device. Device will now get settings remotely from our servers.

Test again to make and receive calls.

Still Problems? Please check our VoIP device troubleshooter

 

NAT/Firewall Issues

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)

linksys nat configuration

 

on the SIP menu;

linksys nat configuration 2

 

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.

Click here for our voip troubleshooter.

Company Profile | Site Map | RSS Feeds | Terms of Use | Privacy Policy | Contact Us

VoIPVoIP™ is a division of Kosmaz Technologies LLC.

Copyright© 2005-2008 Kosmaz Technologies LLC. All Rights Reserved.

  • Kosmaz
    • Kosmaz provides pay as you go voip prepaid phone service and International virtual phone numbers using our own voice over IP system. Bring Your Own Device or BYOD voip plans allow customers to connect their own SIP devices, including IP phones, softphones, and Asterisk PBXs. The solutions are designed for home phone service, small business phone service which can also be used with mobile phones, pc to phone and as phone calling card. We are also sip trunking provider for IP PBX systems such as Asterisk PBX, Trixbox, and any other PBX system that supports SIP protocol.

  • News
    • November 4, 2008

      Whether you're using an open source PBX like Asterisk, Trixbox, Elastix, FreePBX, PBX in a Flash, or an off-the-shelf phone system, softswitch like Switchvox, Fonality, 3cx, Grandstream 5024, Grandstream 5028, Linksys SPA9000, Epygi, PbxnSIP, Allworx, Aastra Talkswitch or a home grown proprietary system, VoIPVoIP SIP trunking service will meet your needs and allow you to focus on your customer, not your network. SIP Trunks provide resellers and business users all of the advantages of VoIP including easy implementation. SIP trunking uses the SIP protocol to allow deployment of voice services over a broadband connection to make calls from 1.9 cents/minute with no volume commitments, no channel restrictions, and virtual phone numbers from 40+ countries or DIDs (inbound phone numbers) from for any US area code.

       

      iPhone voip enables free phone calls now. Download free Fring software in the App Store for your iPhone 3G or iPod touch and configure with you VoIPVoIP account. Within minutes you can make cheap international calls and even free calls to other VoIPVoIP customers with your iPhone.