How to use ast1.4 allowoverlap in sip.conf?

Has anybody understood what we can do with


in the sip.conf of asterisk 1.4(.19.1)? Can we catch added numbers to a sip-number?

Let me us an example: I have a sip-provider which is registered on my asterisk. If someone dials my sip-number (in germany something like +49 228 1234567) he will ring my phone via my asterisk. Could I set it up, so that someone dials

+49 228 1234567 xx

and still reaches my asterisk and there the xx gets proccessed to send the call to an internal extension? If so, how do I catch the added numbers?

I did not find ANY documentation about that configuration. Thanks