Of particular interest are the three analog lines originating at the analog/digital phone switch and terminating at the Digium Wildcard X100Ps in the Asterisk server. If you only have one phone line available (like most homes), only one X100P would be necessary and it would be connected to an analog line running directly to the PSTN.
This setup allows calls to be placed from the soft phones as follows:
- To call another soft phone, dial the soft phone's extension.
- To call any analog or digital phone connected to the analog/digital phone switch, dial 9 plus the phone's extension.
- To call any phone connected to the PSTN, dial 99 plus the phone number.
Incoming calls can be made to the Asterisk server by dialing the extension of any one of the analog lines running into it. The incoming context of my extensions.conf file is configured to direct these calls to three soft phones in parallel. The first soft phone to pick up the call will receive it and cause the other soft phones to stop ringing. If none of the soft phones picks up after twenty seconds, a short message is played and the incoming call is hungup.
Note that the number of analog lines running into the Asterisk server is the bottleneck in this configuration. While the number of soft phone to soft phone calls is limited only by the available bandwidth on the LAN, there is a limit of 3 calls involving a soft phone and any phone connected to the analog/digital phone switch.