gswitz
Yes.. The track gain is set to a useful range rather than complete range.
Clip gain is automateable. Not sure how low that goes. I've never pushed lower bounds on these parameters.
You can mute tracks and clips.

Its not that I ever need to go that far down in level when using the gain knob. I just thought to myself the other night, what if I bring the knob level all the way down? When I did, I noticed the audio was still there. It slightly confused me... But with this post and everyone's responses, I think we have cleared the confusion. 
 
Thanks again all!!!

AT
Best practice is to capture as close to how you want it to sound, at a good level.  That makes everything else easy and leaves "mixing" to adjusting levels and pan and reverb and other such ear candy.
 
If the track's level is too low, for whatever reason, either the gain knob or normalizing can be used to raise it higher.  Either of those have the same effect, raising the entire signal, peaks and noise.  Compressing, well, compresses the internal dynamics of a sound, squishing the differences between the loudest signal let through and the softest part, including noise, esp. if you have to raise the output level on the comp.  Compression evens out the signal differences between the loudest and softest parts of said signal.
 
Normalizing isn't supposed to be a normal, everyday tool to make up for bad tracking techniques, but a tool to fix problem tracks.  That said, if your process involves normalizing everything to make mixing easier since every track as loud (lead guitar, kick and triangle), that is fine, but seems to me the long way around and is a way to introduce unnecessary noise.
 
However, gain staging is mostly an analog problem in driving a signal hard enough to overcome inherent noise without saturating or distorting.  Once it is in the computer you can raise a signal or normalize or compress it w/o needing to introduce any artifacts or adding noise (you'll just be raising up whatever is already there).  For me, if the captured signal needs more level to work into a mix, I'll use gain for a little bump, but normalize when I need more than a couple of dBs of gain.
 
@

This is the analogue approach and does not work for the digital paradigm.  Yes get your levels right before going into the box, however...if you aren't recording analogue, and here is the rub...you need to be aware of both ways depending on your practice.  Aim for -6db RMS on the Sonar meters for analogue recordings, but don't set and leave...just like a cake or a painting, think about the layers.  So for some rock, the vox once it's inside the box should be 0 VU say on the console emulator VU.  Same with drums, but the bass could be -6 on the VU meter.  And if you set and leave, you can't set a mix up like this.  This is what the gain pot is for in Sonar, Sonar is the best mixer in DAW land because of this.
 
If your not sending the audio out of the box after it's been recorded or if its totally electronic, think about the cake and use the gain pots to set a whole mix in the above way.  And here is the important step, don't worry about overs...it does not matter about clipping once your in the box, because of 64FP.  And this is how you make a digital mix sound analogue-fat and warm .  Just ask if you don't understand. 
 
All my work is on my websites, which are in my signature.  I believe my work speaks for itself now.
 
Peace Ben :)
 

John
Really its not a good idea to clip any audio even with a FP audio engine. True the routing within wont have a problem  and with FP there is no true clipping but some plugins can't handle overly hot signals those above 0dB. Besides there is no logical reason to allow audio to clip.
 
A technique I rely on is before I start to mix I lower all faders to infinity. I then raise one at a time to set a balance. If I find I'm in the red on a track to get it in balance I go back and lower all the others. Remember faders go both ways.    

No and take this the wrong way John, but this is old school thinking.  And I'm not talking about constantly in the red either, just don't worry about clipping...just like on a analogue desk.
 
I do the opposite to John, I leave all my faders at zero, mix all the instruments e.g. console emulator, EQ, compression, then balance the mix to fit the quietest track.  This generally means the master bus is over by 3 or 4 db, I then use the gain pot with Linear Phase EQ, then Tape Sim...to give me an average of -3db RMS headroom for mastering.  This is how you use the gain pot. IMHO.
 
But John is right, and it can work this way...although I'm highly doubtful you will get the same fullness of mix this way.  And it is how an analogue engineer would mix in the digital realm.  Whereas I'm a digital engineer, using analogue mixing techniques.  
 
Now there will be some huffing and puffing about this concept, this is analogue engineer vs digital engineer, but as a digital aesthetics expert now...and I will get around to writing a paper and publishing this at some point, there is a definite difference between the two paradigms.  Again the proof is in the pudding, all my music is in the below links.
 
Cheers Ben 

So for the guys who normalize your tracks how are you doing this?

If you track at your chosen rms ref level for each track and you are in 24 bit, no gain adjustments are usually required from that point on.
 
But for whatever reason you may need to add or subtract gain to a track prior to mixing.  (especially and most always if you are mixing a session tracked by someone else. Very few get this correct track rms level thing going on!)  Normalising in that sense just takes the highest peak in the whole file and adds enough gain to the whole track so that peak can reach 0 dB FS.  Which is also not really wanted either.  And it is also a process that factors the peak values in where they are not really important or as important as rms levels.
 
A much better option is measure the rms level of a track and either add or subtract gain to that track  (I usually do that in the editing program)  so that the rms level matches your chosen reference again. This is different to normalisation but a better process for this kind of mix prep.
 
John's approach of starting with nothing and adding stuff in to create a balance is a good one. Even better to that is do this into a stereo VU meter.  The idea is to get the rhythm section pumping to about -3 or -4 dB VU and then by the time you add all the lead elements and other stuff in you have a got a full mix just peaking 0 dB VU.  When all your tracks are at the correct rms level then you can easily attain this situation.  And most often the faders feeding that buss or main buss will all be down a little from unity but almost at a similar setting too.
 
The 0 dB VU ref level remember is down at the digital reference level which may be -14 or -18 or -20 dB FS.
 
That mix will sound huge especially if it has either 14, 18 or 20 dB of transient headroom above the rms level. Turn it up loud in your room and revel in how amazing it sounds.  The rms levels inside your DAW are always related to the SPL level in your room.  When the SPL level in the room is correct you also tend not to push levels up on tracks and buses etc in order to get the volume.  The volume is already there.  In fact when the SPL monitoring level is up then you tend to hold track and buses back a little more and let them breathe etc.
 
We did all this in analog years ago and for me it still applies.  Except then the analog headroom ceiling was a bit like a piece of thin stretched rubber.  Loud things could go up and well over but it was a peak that was absorbed a bit.  In digital our ceiling is more like a thin glass ceiling and we need to make sure whatever we do, still sits a little under it so to speak.  Otherwise we can crash through and the results can be unexpected, usually the signal quality breaks at around 0 dB FS so there is no need to get so close to it really.

If your converters can output a maximum level of +22 dBu then it means at the nominal level eg +4dBu you will need to operate as 0 dB VU as -18 dB FS.  No harm in dropping down to -20 as your reference either.  That just means at -20 the output level leaving your converters will be +2 dBu. Which is still fine.
 
As far as I know Sonar's rms meters are not able to show high up on the scale eg 0 dB VU at the reference level.  The rms meters will always be way down on the scale.  There is also another issue with Sonar rms meters and that is they show 3 dB lower than the industry standard as well.  (they are showing actual rms as being 3 dB down from peak but that is not the agreed industry standard)  The industry standard is that for a continuous tone the peaks of the sine wave should be at the ref level.
 
The best way to overcome the lack of proper VU display in Sonar is to invest in some VU meter plug-ins such as from Klanghelm or PSP.  Both make excellent VU meter plug-ins with a proper VU scale and the ballistics of the meters are close to the real deal too.  They are not expensive either.  You can set the reference level with these plug-ins so if you set the ref to say -20 then the VU will show 0 dB VU at -20 dB FS which is what you are after.
 
http://www.klanghelm.com/VUMT.php
 
http://www.pspaudioware.com/plugins/tools_and_meters/psp_triplemeter/