Question : ASTERISK HELP PLEASE: Not Sure of Correct Terminology For Incoming Calls to Roll To Different Call Path and NOT Ring Busy

I am getting a SIP trunk with 24 call paths from Windstream.  I have an Asterisk FreePBX server.  I don't understand the correct terminology of how I want to carve up those 24 call paths so I am stumped at researching it on forums.  Below is the scenario I will have:
 
1st Main # 865-555-1234
2nd Main # 865-555-4321
Pool of DIDs (which may not matter for this example)
24 Call Paths on the SIP Trunk (meaning I can have up to 24 calls going on at once)

When an outside caller calls into 1st main number I want the call to ring into the receptionist.  If a second outside caller calls into the 1st main number while the first call is still ongoing (and may have been transferred to an end-user)  I also want that to ring into the receptionist (NO BUSY SIGNAL).  And so on....   We are publishing the 1st main number in the phone book and anyone calling in will always call that number normally.  We have DIDs too but don't always give those out and not all phones have a DID anyway.  So basically as long as there is a free call path available I don't want any outside caller to get a busy signal when dialing the main phone number.

The same process for the 2nd main number is true as well but we don't publish this number.  It also needs to ring into a different receptionist and also never ring busy (as long as one of the 24 call paths are free).

I hope all that makes sense.  What I need answered is this .... what is this called in the Asterisk world and can you link to a white paper or something that explains it to me?  I don't need to know the technical details of how to do it just yet ... just what this is correctly called and a decent article/paper on it.  I'll ask a new question if I can't figure out how to get it working after that.

Thanks in advance for all advice.

Answer : ASTERISK HELP PLEASE: Not Sure of Correct Terminology For Incoming Calls to Roll To Different Call Path and NOT Ring Busy

With SIP they can send you as many calls as you have bandwidth for down the same trunk. In your FreePBX configuration under Trunks, you will define a trunk for Windstream and specify 24 simultaneous channels. In your Extensions config, you will define all phones connected to the system. Then in the Incoming Routes you will define a route for every DID number you have coming in. This is where you will define your routing. Create a route with the DID number of 865-555-1234 and point it to the first receptionist. Create a second route with a DID of 865-555-4321 and point it to the second receptionist. Now everycall coming in with 865-555-1234 will go to the first receptionist and so on. If in the Extensions you define a voicemail for the receptionist position, then calls not answered will route to the voice after the ringing times out.

Obviously this is a very high level over view and if you need help with the details let me know. There are companies out there that can do this configuration for you as well. Ours is one of them.