Dynamically Routing Calls To SIPphone Peering Partners
Since we are constantly connecting with new phone networks (peering), you can use our network to route calls to other existing peering partners. This allows instant and dynamic access to our list of other peering, giving you the ability to route your call over our network for free.
Without dynamic call routing, you will never have an current list of numbers you can route for free using SIPphone.
To do this:
- When a call is made send an INVITE to SIPphone (in the form 1XXXXXXXXXX@proxy01.sipphone.com for the number you are trying to dial)
- If the proxy returns a "401" message, this means that the call cannot be routed toll-free by SIPphone and must be routed by traditional means. You will use SER "t_on_failure" route handler to do this routing
- If the proxy returns "100" and a "200 OK" SIP response then your call will be routed toll-free by SIPphone
- All of this back-and-forth takes milli-seconds, so it is completely transparent to the caller
SIPphone is currently peered with some of the biggest phone networks in the world, many universities (UCSD, UCI, etc), other VoIP networks and Asterisk's DUNDi peer-to-peer call network, so dynamically routing calls will increase the cost efficiency and performance of your network substantially.
All of these services are offered for free. If you need any assistance connecting with SIPphone, please contact SIPphone Peering Assistance with your questions.
Get Gizmo5 Mobile Now
Get Gizmo5 Desktop Now
| Download Gizmo5 on your computer. Call for free, IM, and share files. |
 |
| Works on Window, Mac, Linux and Nokia Tablets |     |