]> git.ipfire.org Git - thirdparty/asterisk.git/commit
Add "send to voicemail" Digium phone functionality to Asterisk.
authorMark Michelson <mmichelson@digium.com>
Mon, 21 May 2012 19:05:52 +0000 (19:05 +0000)
committerMark Michelson <mmichelson@digium.com>
Mon, 21 May 2012 19:05:52 +0000 (19:05 +0000)
commit65e49d0ae4bfcc347a2b9722e9d024726086a114
tree75024631308aefdb9502ab36d3909294060fed28
parent3ef6bb1d21bd9fb02f0f4e4b392b2cdbb534f1a6
Add "send to voicemail" Digium phone functionality to Asterisk.

This change accommodates two methods by which calls can be directed to
a user's voicemail.

* Incoming calls can be redirected to any user's voicemail.
* Established calls can be blind transferred to any user's voicemail.

Digium phones indicate the desire to direct a call to voicemail by using
a Diversion header with a reason parameter of "send_to_vm".

This patch adds the "send_to_vm" reason as a valid redirecting reason. In
addition, chan_sip.c has been modified to update redirecting information
on the transferred channel by reading a Diversion header on a REFER request.

(closes issue AST-871)
Reported by Malcolm Davenport

Review: https://reviewboard.asterisk.org/r/1925

git-svn-id: https://origsvn.digium.com/svn/asterisk/certified/branches/1.8.11@367161 65c4cc65-6c06-0410-ace0-fbb531ad65f3
channels/chan_sip.c
include/asterisk/callerid.h
main/callerid.c