VOIP decoding / encoding question

Say I have VOIP setup on an asterisk server, someone calls in via IP, and the answering party is on an analog line (regular telephone) connected to an asterisk server on an FXO port. What actually does the decoding, does the asterisk server handle the decoding to analog?

The IP phone and the analogue line card, for the respective two ends. (The ear, as well as the analogue phone line, is an analogue device.)

Asteisk may transcode, e.g. from G.729 to a linear code.

As Asterisk is software, and therefore purely digital, I don’t really understand the question.

Maybe I have a misunderstanding of what asterisk does or what it can do.

Say I am using an IP phone from a remote location. I dial the IP address of my asterisk server. The server then routes this call to my analog phone which is on an FXO port off of the server.

The VOIP call is encoded correct? So something has to decode the data so that the user with the analog phone can have a conversation. Since my analog phone does not have a (DAC) what will be doing the conversion?

The line card containing the FXS interface. As well as Asterisk, you need a PC with an ethernet interface and a supported line card.

Aha! :smile: That is what I was wandering. It did not know if all that work was offloaded onto the processor or if the TDM400P had a DAC or some kind of chip which handled that…