asterisk users - Jun 2005 - Call being answered, but no audio on either end (Intermittent)

The best type of error possible, intermittent.

We have PSTN numbers being switched to SIP then forwarded to our Asterisk
server which sits inside our LAN

Every once and a while (maybe 1 out of every 20 calls) goes like this:

    -- Executing Answer("SIP/213.199.36.50-0818e3e8", "") in
new stack
    -- Executing Ringing("SIP/213.199.36.50-0818e3e8", "")
in new stack
    -- Executing Dial("SIP/213.199.36.50-0818e3e8",
"ZAP/g1/:8213") in new
stack
    -- Called g1/:8213
    -- Zap/1-1 answered SIP/213.199.36.50-0818e3e8
    -- Hungup 'Zap/1-1'
  == Spawn extension (from-gv-uk, 441252580625, 3) exited non-zero on
'SIP/213.199.36.50-0818e3e8'

Looks normal right? During this whole exchange, neither side can hear the
other. Not even a ringing sound.

The above looks no different than the successful calls.

Has anyone seen this type of behavior before?

Thanks!