[asterisk-dev] "hangupcause" field from "sip_pvt" structure shouldn't have the channel especific hangup cause ?

[Mauro Sergio Ferreira Brasil]

Hello there!

Making some tests and adding lots of logs here I noticed that the 
channel "hangupcause" field gets succesfully updated with Asterisk 
internal hangup condition code when a call (i.e. SIP INVITE) fails.
Obs.: Asterisk version 1.4.26.

The question is: why "sip_pvt"'s same "hangupcause" field is equal to 
"0" on such conditions ?
Shouldn't it be the channel's especific hangup error code (like SIP 
error 486 [Busy], 482 [Loop Detected], 503 [Service Unavailable], etc) ?

Considering that you agree that current codes really don't update this 
field the way I've pointed above and that you guys don't see any 
problems on make necessary adjustments at "chan_sip.c" to make it 
happen, is there any possibility of a patch with such adjustments be 
approved and incorporated on Asterisk trunk?

Thanks and best regards,

-- 
__At.,                                                                                                                             
   _
 
*Technology and Quality on Information*
Mauro Sérgio Ferreira Brasil
Coordenador de Projetos e Analista de Sistemas
+ mauro.brasil at tqi.com.br <mailto:@tqi.com.br>
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