I’ve been using Dialogic gateways for years. I have 4 PRI T1s in them that convert inbound calls to SIP and then send to my back end Microsoft Speech Server. The Dialogic gateways have route tables that basically say “if someone calls 856-555-1111, send the SIP call to A@192.168.10.10; if someone calls 856-555-2222, send the SIP call to B@192.168.10.10; so forth and so on”.
Rather than keep adding more PRIs and Dialogic gateways ($6K each!), I can get SIP trunks from my phone carrier right to me. However, they cannot do the routing I’m currently doing. All they can do is send DID to static IP. I will have to route it in some fashion so the correct SIP address is reached.
Will Asterix do this for me? Essentially see a SIP call come down the line on 1 NIC (out facing) and then send it out on another NIC (in facing) to the correct SIP address like my routing above, based on the inbound DID?
Thanks for any insight and pointing in the right direction. I’ve been using hardware solutions for the last couple of years and a complete software solution is new to me.