it depends on the ATA (analog to IP) adapter that you are using. There are several options:
- if you use a sound card in your asterisk, there is a channel in asterisk based on the linux ALSA framework (I think it's called console). It should be added to AMP using the "custom" extension template.
- if you plan to use a PSTN to SIP gateway, like a sipura 1000 or higher, then all you need is to configure a generic SIP phone. You can add it in LMCE or directly in AMP as a SIP extension. A good resource to know what parameters are specific to your ATA or SIP phone is the site http://www.voip-info.org
- there may be other setups, but I haven't tried. How is your doorbell connected ?