asterisk users - Jul 2005 - SIP calls to 'BUSY' or OFF HOOK PSTN numbers do not return busy indicate to sip phone?

What we would like to see happen or emulated is that if someone
makes a call via our SIP provider to a PSTN number that is actually
busy that we get an actual BUSY tone at the telephone.

In our test case this is a PAP2-NA SIP device

It would appear that when we call the far end (PSTN phone number)
that is busy we do not get any busy indication at the user end (originating
telephone on our Asterisk system) that the line is actually busy.

Asterisk 'knows' it's busy/congested but does not seem signal the
PAP2
of this condition.

If I do a test directly on asterisk like this:
exten => 7777,1,Answer
exten => 7777,3,Busy
And call extension 7777
It sends a 'busy' to the PAP2 and we hear the normal
audible US busy indication.

We would like this to happen when a user actually calls a PSTN
telephone that truly is 'busy'.

What actually happens at the originating phone is:
User dials PSTN number (which happens to be busy/in use)
User hears about 10 seconds of dead silence
User hears "Goodbye.... Thank you for trying the Asterisk Open SOurce
PBX"
Hangup is executed.

This seems to be the 'default' behaviour of Asterisk and also
does not seem to be right :-)

Here is what asterisk shows on the console when a normal BUSY number
is dialed:

    -- Executing Dial("SIP/4077-030d",
"SIP/17346621122@sipprovider-out")
in new stack
    -- Called 17346621122@sipprovider-out
    -- Got SIP response 486 "Busy Here" back from 216.164.114.122
    -- SIP/sipprovider-out-c3fd is busy
  == Everyone is busy/congested at this time (1:1/0/0)
    -- Timeout on SIP/4077-030d
  == CDR updated on SIP/4077-030d
    -- Executing Goto("SIP/4077-030d", "#|1") in new stack
    -- Goto (sip,#,1)
    -- Executing Playback("SIP/4077-030d", "demo-thanks") in
new stack
    -- Playing 'demo-thanks' (language 'en')
    -- Executing Hangup("SIP/4077-030d", "") in new stack
  == Spawn extension (sip, #, 2) exited non-zero on 'SIP/4077-030d'