Help.
Why is it that I can call out from my GSBudgetone SIP phone but the audio is "one-way'?
Why is it that when I call my asterisk phone number, I get a fast busy?
Can we have a looksie at your config files? sip.conf, extensions.conf, zapata.conf, zaptel.conf.
If you start asterisk with enough verbosity:
asterisk -vvvvvvgc
what does the console say when you make an incoming call to your asterisk box and get the fast busy?
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