[asterisk-dev] SIP disconnect code

[Johansson Olle E]

As I said, this will require quite a lot of work, but it seems like  
we're getting a concept outlined here.

I would like to add that we need a callback function to the channel to  
be able to aggregate codes
(if multiple channels are involved in the call) and suggest one result  
code. For SIP, there are rules on how to
handle this in the proxy part of the RFCs.

Also consider that the text after the code varies quite a lot, you can  
not rely on the text after the SIP
result code for any parsing or logic. It could be in Swedish, Chinese  
or even English....

/O