Just starting with Asterisk

Asterisk Asterisk Support
1

Ok, here goes. I’ve pretty confident with a computer and I can learn just about anything. I’m asking for help/pointing me in the direction of where to go to get Asterisk up and running on my WZR-HP-AG300H with DD-WRT and Optware installed on it.

So far I have gotten Optware installed, and installed the package for Asterisk 1.8 (opkg install asterisk18). So I assume that I now have asterisk installed. I have logged into the router using SCP and transferred the following files from the router: modules.conf, sip.conf, asterisk.conf, extensions.conf. At this point is where my confusion starts, specifically when I looked at the sample configuration for SIP.

Here is what I’m trying to do. I have 4 Mitel 5330 IP phones with SIP enabled on them. I have OOMA home service, and I have a Cisco SPA3102 in the mail and on the way to me. At this point I just want and Asterisk server configured that will do the following:

  1. Connect to my Mitel phones via SIP (all on my home LAN i.e. not from WAN)
  2. Connect to SIPDroid on my Android phone (Need to be able to connect from WAN)
  3. Allow me to dial between the rooms at my house using an extension
  4. Receive incoming calls and make outgoing call from the OOMA -> SPA3102 -> *
  5. Keep Caller ID
  6. Need it to be secure

Those are my most pressing concerns and if I had that working I could live with not being able to do anything else.So if anyone can help me with this I would appreciate it. On this site I’ve done a lot of browsing and all I keep seeing is info on Google Voice, while I have Google voice and think that would be nice. I don’t need * for this. So any help in my above setup or pointing me in the right direction I would extremely appreciate.

The below are my “I would be very grateful if they worked” list:

  1. Ability to page via speakerphone all the phones in the house
  2. Ability to specific calls
  3. Ability to change Caller ID info (I’m calling from work…boss)
2

You should start with Asterisk built from source on CentOS or RedHat, on a desktop, laptop or netbook. You should start with phones intended for open SIP use, not ones designed for a proprietary system that just happens to use SIP. We use Polycom’s. Digium are about to launch one under their brand… I would suggest at least two such phones, and not softphones.

Although it is trivial to get Asterisk to send any reasonable caller ID over SIP, in my view, any SIP gateway service that allows you to send a caller ID that you have not proved to them that you have the right to use is probably a rather dodgy operation. A respectable one might forward one, but should flag it as “unscreened”.