[Asterisk-Users] Call being answered, but no audio on either end (Intermittent)

[Geoff Manning]

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!