Add as a Voice Service Provider for quick configuration

Started by BCITMike, March 08, 2013, 12:20:34 AM

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The Google voice setup is quick and easy because you have pre-configured some settings.  You would sell more Obihai's in Canada if you added FreePhoneLine to the Voice Service Providers when adding a new account.

Here is a link with the recommended settings:

VoIP Unlock Key Credentials

posted this on Feb 07 12:53
Version 1.01

The Freephoneline VoIP Unlock Key provides sip credentials that can be used to configure any SIP client to work with the Freephoneline service. Please review the following Freephoneline guidelines to set up your SIP client. Failure to follow the required guidelines will result in account suspension followed by a notification email. Once your configuration adheres to the guidelines, service will be restored.

These guidelines may change over time - if and when they are changed freephoneline users will be notified at the email address used for account login a minimum of 7 days prior to the changes being required. Please ensure your contact information is accurate by visiting your account profile on

Required Settings
SIP Server:

Alternative SIP Server:





It is always best to use the DNS name for your SIP server as our infrastructure is always expanding/changing/being maintained. The IP addresses which you register to will change over time.

Use of Fongo SIP Servers that are not listed in this document will result in your account being suspended.
Registration Interval:

3600 seconds (1 hour)

Registration Expiry: 

3600 seconds (1 hour)

Failed Registration Re-Try Interval:

120 seconds

Recommended Settings


NAT Mapping Enabled:


NAT Traversal:

Enable sending Keep-Alives only:

on Grandstream HT-701 ATAs this setting is "no, but send keep-alive"
Keep Alive Message:


For Linksys/Cisco devices, use 'Nat Keep Alive Msg' = $NOTIFY or $PING
Never use REGISTER as your Keep Alive message
Keep Alive Interval: 

20 seconds*

*Audio may be affected if this value is adjusted

The above settings can be used to configure your SIP client to function in common home network configurations. Since there a thousands of home network configurations, it is impossible for us to provide a single set of parameters that will always work. As a VoIP Key purchaser, it's expected that you have knowledge of your network and how to configure your SIP client properly.

Freephoneline does not offer STUN server. However, you may use a public one if your wish.

RTP Settings
Supported Codecs:

g711-u (uLAW)  and g729

Suggested RTP Packet size (psize):

0.020  - This ensures audio packets every 20 milliseconds, achieving better quality (trade-off: bandwidth)

The above settings are used by your ATA to determine how the audio will be encoded/decoded across the Fongo network.

Additional Information for users with multiple SIP clients on their network
If you use multiple VoIP providers or SIP clients, including Dell Voice or Fongo Mobile on the same network you may encounter issues if your router does not support UPNP.



I believe that the various service providers need to pony up in order to be on that easy list. That's just one way they monetize the product/service.

If more people make noise maybe it might get the attention of FPL to request a listing.
Owner of the 1st OBi110/100 units in service in Canada & South America. 1st OBi202 on my street. 1st OBi1032 in Montreal.


I also would like to see  Freephonline on the list.  I can't seem to find where to enter the alt sip server.