OK, I’m still somewhat of a newbie myself, but let’s try to get some more info.
Is your softphone either  or  in sip.conf?
You can confirm you’re registered with broadvoice with “sip show registry” at the CLI. Have you checked this?
What number are you dialing (you don’t have to post the number, but show something representative so we can see what pattern it matches in the dial plan).
Is there anything else in the console output (besides what you posted in your original post)? Are you running with a high level of verbosity (e.g., asterisk -vvvvvvvvr)?
For example, when I make a call to telasip from my ATA-attached analog phone, I get output something like this (I substituted xxxxxxx for the actual number):
-- Executing Dial("SIP/john-desk-8db7", "SIP/925xxxxxxx@telasip-gw|60") in new stack
-- Called 925xxxxxxx@telasip-gw
-- SIP/telasip-gw-a323 is ringing
-- SIP/telasip-gw-a323 answered SIP/john-desk-8db7
-- Attempting native bridge of SIP/john-desk-8db7 and SIP/telasip-gw-a323
== Spawn extension (custom-callboth, 925xxxxxxx, 5) exited non-zero on 'SIP/john-desk-8db7'
-- Remote UNIX connection
The “SIP/telasip… is ringing” and “SIP/telasip… answered” messages show call progress. When the call is answered, it is bridged, as shown in the Attempting native bridge.
Comparing this to yours, it seems like you’re getting connected (you get the attempting native bridge message). Is it after this message that you get the recording?
Is it possible the target number on the PSTN has some problem? Can you call it from a regular land line? Can you call a different number through asterisk and get the same recording?