[asterisk-dev] SIP disconnect code
Johansson Olle E
oej at edvina.net
Tue Jul 22 02:26:27 CDT 2008
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....
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