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 <[EMAIL PROTECTED]>
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