[asterisk-dev] Extended dial status

[Anton]

Looking at the return of the DIAL command, the return set of 
DIALSTATUS is very limited (just a few states), though 
channels returns quite a rich set of the states of the 
call. For example while using DIAL it's impossible to 
distinguish between SIP return statusses

Got SIP response 603 "Decline" back from x.x.x.x
    -- SIP/sipproxy1-7282 is busy

and regular busy

many other states are not passed too.

That would be hugely comfortable for providing the end user 
proper routing management in real world (to have a VoIP 
call retryed on another termination gateway for instance) 
and also appropriate voice response, why their call was not 
answered, if there would be a way to analyze plain channel 
response, and not only a DIAL version of it. Looking over 
the doc of asterisk i did not found that possibility. 

Does anyone aware of the presence of such a variable, or 
it's necessary to patch every channel to set appropriatelly 
extra variable?