Confused about SIP Trunks/DIDs...?

Ok, so I’m a newbie (obviously), and I’ve just setup my first FreePBX box and did some test calls with a using a softphone, seems to be working so far.

My question is, let’s say I had 2 DIDs at, how does Asterisk know which # to call from? Because when I enter the trunk information I just, basically, enter my account info (username, password). How does it differentiate between DIDs? Because I’m assuming you can have more than one DID# per account?

I know this maybe an extremely n00b question, sorry in advance!


DID (DDI) means direct inward dial, and is for routing your inbound calls, for outbound you have “outbound routes” and trunks. You can select the outbound CID also in the extension as well as the trunk.

yes so my outbound routes, what stops multiple extensions from using my account? or can it support multiple outbound calls?

Think of a “trunk” is a collection of one or my “lines” (old PSTN thinking) Depending on your VSP you might have access to one or more concurrent calls, either inbound, outbound or blended, you might need to check with them what you bought.

hmm… so a trunk doesn’t necessarily mean a PSTN line? I guess I’m thinking the old way… so a trunk is basically my connection to whatever VoIP provider I have an account for? how do I know how many maximum concurrent calls I can have on that trunk?

Yes, the trunk is the virtual facility the calls go across.

Limits and charges and set by the carrier and mostly driven by marketing concerns not technical limitations. Theoretically there is no limit on how many calls can be on a given trunk. Remember SIP is just signalling. Each call has it’s own media path via a protocol called RTP. Theoretically this RTP doesn’t have to touch the Asterisk box, it usually does though because it is challenging to set the network up to do true redirect.