[asterisk-users] Automated Call Testing - end-to-end - SIP Provider

[Positively Optimistic]

We, along with a lot of other people, have a phone number that is pretty
important to us.   Yesterday, our VoIP provider went down...   won't call
any names VI, but it was pretty bad...


Our goal is to create a script within asterisk, that will place a call out
one SIP trunk provider (not the one that provides the DID, and have the
call come back in on another trunking provider (with a special caller-id of
course), and answer it.    If that works, great..  we do nothing.

If the call fails, we generate an email, letting everyone know that our
special provider has went down, again.

We were attempting to do it with .call files, but, for some reason, the
channel variable dies post-call and we can't recording the ${dialstatus} or
use it for logic...


Has anyone done this...?   ...willing to share dial-plan, scripts, etc ?
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