[asterisk-users] SIP failover between Sip Providers

[Alex Balashov]

On Wed, 18 Apr 2007, Knud Müller said something to this effect:

> Hi all,
>
> lets say I've registered at several Sip-Providers. Provider A offers best 
> rates but is often too busy to get a line. Sip Provider B is stable (but 
> more expensive). The asterisk box has a high call volume therefore 
> problems at provider A will get obvious after a few calls stalled. In 
> this case astersik shall switch temporarily to provider B but shall test 
> periodically for selected calls if provider A is available again. I think 
> it can be done by using the dialplan and the database to store the 
> statistical information but maybe there is an easier way that integrates 
> better with asterisk!?

   Best way to do this in my opinion is to deputise this logic to a SIP 
proxy and have Asterisk trunk all of its calls through that.

--
Alex Balashov <sasha at presidium.org>