[asterisk-users] how to use qualify times to route calls

[Matt Riddell]

On 3/03/11 11:29 AM, sean darcy wrote:
> I'm using 1.8.3, and have 2 sip providers. Both are set with
> qualify=yes. Each of them sometimes have qualify times 10+ times the
> other. For instance, one will be at 10-15ms, the other at 200ms.
>
> Is there a way I can route an outgoing call to the provider with the
> lower qualify time?

Traditionally you'd use a value you consider to be good enough for calls 
and set qualify to that.  I.E. if you think 30ms is ok then set 
qualify=30 and then just route via the first then the second depending 
on status.

-- 
Cheers,

Matt Riddell
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