Automated Call Testing – End-to-end – SIP Provider

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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 ?

One thought on - Automated Call Testing – End-to-end – SIP Provider

  • 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.
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