So I'm guessing this isn't possible without hacking the source code, which I've already done to solve my problem for now. But I'd like to make sure there isn't a better way of doing things, and thus I'd like to revise and restate my question for clarity. First, though, let me describe what I am doing, because I think it's a not-uncommon scenario that I think would be something that others may want to do. I basically want to use FS as a simple transcoding server between two endpoints, call them A and B. Calls coming in from A will be using speex codec and I want to send them out to B using PCMU; calls coming in from B will be PCMU and I want to send them to A using speex. The FS server will be a B2BUA and will be doing transcoding only -- no authentication (and no registration). Simple, right? The only fly in the ointment is that A is authenticating calls with B by providing a Proxy-Authorization header. So I need to take the Proxy-Authorization header received on the A leg and include it on the B leg. So far, the only way I have found to do that is to hack the code to create a new channel variable. I've done this, and it works. However, this leads me to the following questions 1) Is there a better way to do this? If there is no way to do it as a dialplan out of the box, can it be done as a script? 2) sofia has parsed all of the sip headers on the incoming invite for us, and they're all available from mod_sofia. Shouldn't those all be available to us (i.e., application developers) by some means (i.e., channel variables)? Dave