SIP to SIP routing

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@; if someone calls 856-555-2222, send the SIP call to B@; 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.

Can you please give some additional info about your request? SIP and DAHDI call routing is more or less the same, so I do not understand, what is bothering you. Call Routing logic is the same, you just use different Channel.

If you want to know, if you have the influence through which NIC you route the call - you do that in Linux routing tables. Asterisk’s job is to route the call and send it to the correct IP address. Asterisk creates the SIP packet and lets the Linux IP stack to do the rest. So the decision from which NIC a packet is sent, is made in the Linux routing table.