I am wanting to do two different things and want to make sure I understand the way ATAs convert to SIP using FXS or FXO ports.
Currently I have an Asterisk Box using Elastix 4.0 (Asterisk 11?).
I want to use 2 pots lines as the outgoing lines. I bought a Grandstream HT703 which has 2 FXS ports. Because I am learning, I also bought a Cisco SPA112 that has [ 2 FXO ports. ] (Oops it has 2 FXS Ports, now I understand I need 2 FXO ports to do what I want.)
1) Which device would I use to connect to the active pots lines? (My current knowledge says the HT703) 2) Do I setup the device (based on question 1) as a SIP extension to asterisk? 3) What settings in Asterisk do I set for using these as outgoing lines? (Preferably in the FreePBX GUI or Elastix GUI, but I will take ".conf" settings if that's the only way to do this)
For a little more information, I successfully have multiple extensions with both NAT’ed remote SIP clients and local hardwired phones/devices. I also am using a raspberry pi as a paging phone and even got the SPA112 working as a SIP client to my asterisk box to make my analog cordless phone system work with the PBX box.
I currently have a SIP trunk with Google Voice (simonics.com) that is working very well. The next step in my testing is to get the landlines working for failover & just quality testing.
Thanks so much for any help.