[asterisk-dev] Detailed SIP disconnect reason

[Kirill 'Big K' Katsnelson]

 From looking at the code, I do not think such a feature exists. If I am 
missing anything, please correct me. If not, them my question is how to 
implement that so it would be of interest to a wider audience.

I need to know in the dialplan why (narrowly, a SIP) call was 
disconnected. In particular, immediately I want to distinguish a 
disconnection from RTP timeout or SIP session timer from another cause. 
But, more generally, if I were to create such a feature, I would also 
include other indications such as e. g. which party has issued a BYE.

So, if I am implementing that,

1. Is that thing of a common interest and may go into the trunk as a new 
feature?

2. What is a better way to report that to the dialplan:
    a - set a channel variable
    b - report that through the CHANNEL() function.
    c - other?

Thanks,

  -kkm
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