Asterisk, conferences, DID, SIP

I’m a little familiar with Asterisk. Years ago I made an asterisk box with some analog cards and accepted incoming calls etc… nothing complicated.

Here is my now project in a nutshell:

I’m trying to create a conference room that an admin can call (regular phone) into and “feed” the audio (speak) so that the other callers can listen (calling in with regular phones). Here is my question: Do I only need to have DID forwarding to my asterisk server. Is SIP trunking required? I will never have to make an outgoing call and will never have a SIP phone connected to the system.

Any guidance would be greatly appreciated. Also, any suggestion on a good DID provider for Canada with good support for AsteriskNOW (freepbx) would be appreciated.

– Sean Fournier

You can try with , buy a DID (1888) then setup the sip trunk with and create the conferance using confbridge