Can I use Asterisk?


My condo provides some kind of VoIP thru PBX for us. I have no idea what they use. I simply connect a regular analog phone to our phone jack and use it like a normal analog phone. The fee is $80 a month with unlimited US & Canada, and free 4 hrs International with $0.02 for each additional international minutes. I am pretty happy with the service because I make around 4000+ min/month. However, the service does not have call forwarding nor call bridging, etc. If I add an Asterisk@home, can I get call forwarding and Bridging? What I want to do is to forward calls to my cellular phone when no one is answering the call at home, and allows me to call from my cellular phone to my home phone, and than dial the outgoing number through my home phone (bridging). This way, I can get unlimited incomg/outgoing cellular minutes too with Sprint’s unlimited mobile to home.

If Asterisk can do this for me, what equipments other then a Linux system do I need? Do I need a HW card or I just need a Ethernet port? Do I need to find out my condo system’s SIP or something?

Please help, thank you.

that price is a bit steep…

anyway, you are limited by the fact that you plug in a normal phone. That’s a POTS (plain old telephone service) analog link, and it can only support one call at a time, doesn’t matter if it’s to * or a phone.

If you were going to link * in, and do things like conference bridging, you would need to have more than one call channel. Either you’d have them install a 2nd line (possibly another $80/mo) or be able to link your server to theirs via SIP or maybe IAX if they use * too. However you would need their cooperation to do this- you would need to get the SIP login/password from them for the * server to use. They may not want to give you this, for that exact reason (that you can use more than one audio channel at a time). Two calls = two channels = costs them more.

Either way, you’ll probably have to talk to whoever deals with your system, so I would suggest give them a call. See if you can get SIP login/pass/server/realm, if not then ask if they can setup a 3way call feature.

Sorry i dont have better news.