I had the same or similar problem with the stable release of 1.4. I ended up having to specify canreinvite=no in my sip.conf so that Asterisk would not drop out of the call path and thus “hear” *1 and start recording.
From what I gather, having ‘w’ or ‘W’ in the DIAL statement is supposed to let Asterisk know not to do this, but I could not get automon (or any other dynamic features) working for me until I specified ‘canreinvite=no’ I am currently testing * and all my extensions are on the same subnet (for now.) It’s my understanding that when start taking/making calls outside of my local network, this will not be an issue.
Speaking of which, does anybody know if there is a way to specify this globally in sip.conf, rather than per extension?