|
I'm setting up a SIP trunk in a voip lab using Cisco CME as my private PBX and Asterisk as the ITSP PBX. This isn't a real world installation, but I'd like to mimic one as best as I can.
My question up front is this:
In a SIP trunk scenario, does my private PBX actually register each phone line with the ITSP, or does call routing rely on SIP redirects and/or call forwards?
In other words, should my Asterisk show several SIP aliases registered at a single endpoint (CME)? |