Normalizing multiple tracks results in a combined wave file - SOLVED!

I searched the forum for this and found the exact same issue, so I know it's not just me...  However, there was no resolution to the issue, so here I am.  When I select more than one track to normalize, the resulting Edit(n) edited wavefile has a W64 extension, and is a combination of the files that made up the selected tracks.  I discovered this when waveforms failed to redraw, even after clearing the picture cache.  The project is pretty big, about 2.5 hours, but I don't know if that has anything to do with it. 
 
I am using X1 build 425.  I have X2, but it seems that X1 is more stable on XP (SP3), and I'm not ready to rebuild a work station that's been configured to my liking over many years.
 
I really need some help on this, as I had no control over the levels I got from the FOH guy, and most of the 48 tracks x 3 days of concerts are low.  I normalize so my compressors, etc standard settings work pretty much the same across various material.  Don't want to argue this point...
 
Thanks.

Hey Jonesey,
 
I looked in the Help/About, and the only version identifier is the build number (425), but I don't think it is specific to a version, as the other entry in the forum was for another version altogether.

Another update.  It does this in X2 as well.

Anyone else?

In my humble opinion you would never Normalize multiple tracks anyway. As what Craig just said. If the goal is that all your audio tracks at say -6Db then each track will need treatment. 
I Tool copy into wave lab to do this action as I find Sonars Normalizing is working blind.
Your better off just using the Gain if  track is low.  

Here are a couple of reasons to normalize, when, due to things beyond your control, the FOH guy gives you levels that are barely there (but in a 24 bit world are still useable):
1. DAWS are a audio-visual tool.  What's easier to work with when you have 2 hours worth of 48 tracks with weak signals, flat lines or visible waveforms?  Consider how difficult it is to replace flubbed notes by the horn section with a snip of a good performance from the same track, if you can't see them...
2.  I, like many others, have mix templates with all of my dynamics processors and other plugins for drums, guitar, bass, keys, horns, etc. already in place.  When levels are way below what you would normally record at, those devices must all be adjusted to sometimes extreme degrees to get proper amounts of compression, etc.  I am talking about situations where channel gain can not make up the difference.
3.  When you have three shows to pick and choose the best performances of in very short order, #1 and 2 become indisputable.
4. Never say never (as in "I will never apply fill-in-the-blank" to my work), (and never tell the FOH guy that he is an idiot - it does no good)

Coffee on standby?

Thanks for your help, Mr. Anderton.

johnnyV
First I like to see what the peak level is, because sometimes a very quiet track will report in a 0db. So now you need to look for that one spot that is peaking etc.
 

 
If you've played the clip, you can right-click on the peak numeric value in the track view and select "go to Peak." The indicator will be a little late compared to the actual peak due to framing, but it's usually obvious where the peak is if you look about 40 ms to the left.
 

Also Sonars Normalizing works with percentage which is totaly weirdness when your thinking db's. Mabey they fixed it in x3 as I admit I haven't bothered with it for 3 years. Tool copy works fine.

 
You can enter the normalization ceiling in dB or percentage. 

I joined this post to the original thread and modified the thread title.
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It does seem unusual that the program can't apply normalization to each clip sequentially or whatever. I don't know much about what's involved with batch processing, but it does seem to be a feature with higher-end digital audio editors but not lower-end ones...maybe it's difficult to do for some reason.

Sony Sound Forge will do batch import normalise and export. Very simple and done much faster than real time of audio being processed.