Increasing the reliability of VOIP telephony with Asterisk

Hi @all,
i have a question about increasing the reliability of VOIP telephony with Asterisk.

What’s set up and working now:

  • each phone has a SIP account with an external operator
  • all external SIP accounts are replicated to the local Asterisk server
  • a rule is set on the router to check if the operator’s external SIP server is available
  • when the operator’s server is down, traffic goes to the local Asterisk server
  • when the operator server is up, the traffic switches back

It works fine, but I wonder if it can be done on the Asterisk server itself.

Best Regards,

You mention the phones have an external service; I assume they have a VoIP account with a provider that gives you SIP logins.

You do not mention how your Asterisk is interconnected to the network. Having your phones just fall over to Asterisk without some sort of trunk to the outside world doesn’t accomplish something.

What you should likely do in this case is you get a SIP trunk from your provider, connect your Asterisk up to them, and have your Asterisk handle all the calls.

I found this strange too. In addition to what has already been asked, I think you should explain the failure modes that you consider important, especially as you seem to be wanting to make Asterisk, or at least the machine on which it is running, a single point of failure.

Also please consider the use of server, it may confuse your thinking. It isn’t been used in the correct sense for SIP, so the only interpretation of it that I can come up with is: the machine on which Asterisk is running. The Asterisk software, itself is a daemon which acts SIP client, SIP server and SIP registrar, at different times.

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