[Asterisk-Users] Outbound Call & SIP Results

[Olle E Johansson]

David Brazier wrote:
> We make outbound calls via our Asterisk (*1), then via SIP to a 3rd 
> party Asterisk (*2), which then routes to PSTN, again via SIP.  If the 
> called number is invalid or out of service, *2 gets a 404 Not Found, 
> which seems appropriate.  However, *2 then passes on a 403 Forbidden to 
> *1, which is not really the right response.  *1 then returns a 486 Busy 
> Here, which also seems wrong, as it would generally mean the called 
> number was busy (engaged), I think.
>  
> Is it possible to vary this behaviour?  The ${DIALSTATUS} variable 
> doesn't seem to be fine-grained enough to help.
>  
Would it be possible to see som log files?


/O