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? Thanks!