Asterisk and SIP trunking, is this configuration possible?

Mike asked:

I am new to asterisk and before I dive in, I just want to make sure that what I plan to do is possible/correct.

My office will run an asterisk server and have both local and remote extensions. We have few people scattered around the US and want something scalable if that number increases.

I have installed asterisk as a VM on VMware ESXi 5 but have not done any config.

If I understand this correctly, I can get SIP Trunking service (the particular one I was looking at provides 1 DID and 5 ports) and have asterisk use that as the POTS gateway for outgoing calls. This will allow any extension to pick up the next free outgoing line if they want to make a call (right?). Is that a function of the SIP trunk provider or Asterisk?

For incoming, we are already using twilio, so I was planning on keeping that since they now have SIP routing. So I assume I can use their call tree and route to my asterisk extensions. Can I duplicate twilio functionality in asterisk?

My answer:

Your provider’s sales droid really should have answered most of these questions for you, but…

It’s your responsibility to configure Asterisk to route calls in the manner you want, but the sort of routing you want is certainly possible and not very difficult. Though trying to reimplement all of Twilio would be a major project, just setting up the specific routes you want isn’t very hard.

If you have 5 “ports” this generally means you can have 5 simultaneous calls via that provider; a sixth call would have to wait or be routed to another provider. Ask the sales droid for clarification on whether this covers both incoming and outgoing calls; sometimes they are billed differently.

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