asterisk users - Nov 2013 - Automated Call Testing - end-to-end - SIP Provider

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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It's look like our test Did script, which was test a list of our did
number. it generate call files which call to a number and, after answer
play file for a 4 sec. After this, we send email for manager with excel
file where everely Did number noted mark. like this. sorry for my english.
08 ????. 2013 ?. 23:38 ???????????? "Positively Optimistic" <
positivelyoptimistic at gmail.com> ???????:
> 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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