[asterisk-dev] Asterisk 1.8 and SIP Diversion: header

[Pavel Troller]

Hi!
  Since 1.6 days, my dialplan is handling adding the Diversion: header to the
set of SIP headers for a diverted calls. The diversion is handled internally,
by the dialplan itself (using the database query) and I don't know a method,
how the destination channel (in this case SIP) can get a diversion information.
  Now, in asterisk 1.8, during tracing SIP calls when searching for another
problem, I found that now I have TWO Diversion headers. One is my own old one,
and the second seems to be added by the SIP channel automatically (and I really
don't know where the channel is getting this info).
  The problem is, that the header added by me is better: it contains a reason
information (the second contains reason=unknown) and the screening, privacy and
count information (the second one doesn't contain neither of them). I would
like to solve this problem by:
  1) either supplying the missing information (reason, screening, privacy) to
the SIP channel to be able to build the header properly and stop adding my one,
  2) or keeping to add my one in its full glory and tell the SIP channel not
to add its one.
  However, I don't know, how to achieve neither 1) nor 2) (with other means
than directly patching the source code). Maybe, it would help me to know, how
the channels gets the info that the call is undergoing a diversion, to mask
it out of the channel, thus to prevent it from adding the header.
  
  With regards, Pavel