[asterisk-users] SIP failover between Sip Providers

[Knud Müller]

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!?

regards,

-- 
Knud A. Müller