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

[sean darcy]

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?

sean