Asterisk voip only setup

Could someone tell me if this is how it is done?

  • Purchase a DID number, have it sent to my static IP. Let customers know what the number is.

  • Setup a AsteriskNOW box to react when it sees a call coming in on that number.

  • Purchase an ATA devices so my five multi-phone Panasonic base station can ring when a call comes in.

  • Purchase the Skype module so I can dial out using my Skype username.

Does anyone still use Asterisk anymore? I thought it was hot.

What you described makes a lot of sense.

Yes it makes lots of sense, but you really shouldnt expect us to walk you through it. You should read:
That will tell you alot of what you want to do, plus it will give you a little background in telephony. I suggest you read that then come back with more specific questions on your setup. Good luck to ya, and we will be here but we just cant walk everyone through setting up a system and making it work for their specific scenerio…

Thank you gentlemen. I have been reading and learning a great deal. DialPlan Syntax is kicking me. I have also been working with PBXnSIP. I really like the flexibility of Asterisk however. This is what I have to make work, I know you guys are going to make me do it myself, but… perhaps a generous soul will post. The Asterisk community could only benefit if I get this working.

Call Flow:

  • Caller dials 555-123-1234.
  • IVR sees it is 9:00-18:00 and plays “WelcomeToCompany.wav” else plays “AfterHours.wav” and sends to voicemail.
  • Silently connect caller to FindMe/FollowMe ring group which consists of extension 2000 and external cell 555-222-2222.
  • Play sound “MoH.wav” while the ring group does it thing.
  • Extension 2000 rings 6 times, then cell rings 6 times, if cell answers play “Press1ToAccept.wav”.
  • No extension answers so play “AllBusy.wav” and send to extension 2000 voicemail.

To summarize:
The caller hears an auto attendant, they press a button or wait, they hear music, then they hear a live voice or go to voicemail.

that is a real basic callflow and you should have no problem with it, it is also described in the book above…