[asterisk-users] SIP ATA Channels for outbound calls - How to select in dialplan

[Dean at INKnBITs]

I have setup 3 Linksys SPA-3000 devices to pass/send our analog voice calls
into/out of asterisk. The inbound calls work fine as I have set the
spa-3000's to forward all calls to an extension. I have added them to the
sip.conf as spa-3k1, spa-3k2, and spa-3k3. Is there a way for when some
picks up a phone to dial, it starts at 3k1, if congestion, move onto the
sk2, and so on. I'm looking for it to find the first available line to use.
Is this possible in the dialplan?

Thanks,
Dean.