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

St_Dwarf stdwarf at gmail.com
Fri Nov 8 13:54:17 CST 2013

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 нояб. 2013 г. 23:38 пользователь "Positively Optimistic" <
positivelyoptimistic at gmail.com> написал:

> 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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