Digium phones and FreePBX


I’m using :
Asterisk 11
FreePBX 2.11
Digium Addons

Everything works fine once you set EASY MODE to NO.

My question is about what “more” can you do with Digium Phones.
We mostly have the D40 model.

We can do a lot of cool stuff, set the logos, ringtones, etc. from FreePBX but I feel you have to be able to do something more!

For example:

A) There are two “line-keys” on the phone, is it possible to do anything with that? For example, set each line-key to a different ASTERISK outbound route? Or anything else?
The only thing that seems to be possible is to configure an external line on a line-key… which is quite useless…

B) Is there any way to program custom features (program feature codes, etc.) on those phone? If so, how?

Also, a question, I’ve noticed that when adding a phone, and associating an ASTERISK extension:

It seems to be doing a difference between INTERNAL and EXTERNAL extensions.
However, I don’t see anywhere in FreePBX where I can manage that (setting an extension to be external).
Any ideas on this?


A) Currently, that second key can either be a second internal SIP registration (extension), a registration to some other system (a second line), or it can be mapped to a SIP URI (speed dial), with (BLF) or without subscription.

B) Like what? Use the second line key for that.

C) Are you talking about the phonebooks? We’re differentiating between extensions that are local to the system and extensions that aren’t local / are some custom thing that isn’t represented as another user on the system.

Thanks for response…I am having similar trouble finding out how to use the 2nd line as a separate external line. IN other words the first line is my extension and the second is an outside line where i can place outside line calls via the same trunk that is setup with sip provider where i have several dids. I have been told i can do this without setting up multiple trunks within freepbx. can i only set these other lines (D50) up as extensions within free pbx? thanks for help.

There is no such thing as a line in SIP so I am not sure what you are trying to say. You can assert any valid CID on an outbound channel with your SIP trunk.