res_pjsip_send_to_voicemail: transferring to voicemail for digium phones
authorKevin Harwell <kharwell@digium.com>
Tue, 25 Feb 2014 17:47:06 +0000 (17:47 +0000)
committerKevin Harwell <kharwell@digium.com>
Tue, 25 Feb 2014 17:47:06 +0000 (17:47 +0000)
commiteee4313fe8c1563e64d557c46d8e070275d07e99
treebcb3c69e1fa59c96cd584b251194f57d6474e6e2
parent23b142d5c8ca6c9d04e2a71be9dfbc1ff4c1527e
res_pjsip_send_to_voicemail: transferring to voicemail for digium phones

Added the ability for transferring directly to voicemail on digium phones.
Added a new module that checks for the presence of a custom header and/or
diversion header within a sip REFER.  If either is found and they specify
a sending to voicemail action then variables are added to the channel
allowing the user access to them in the dialplan.  Dialplan can then be
written that branches based upon these values allowing, for instace, for
a single number to be used for dialing and/or accessing voicemail directly.

Also fixed a problem where the PJSIP_HEADER function was allowing non pjsip
channels through (checked to make sure it has the correct channel type before
proceeding).

Review: https://reviewboard.asterisk.org/r/3245/
........

Merged revisions 408880 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408881 65c4cc65-6c06-0410-ace0-fbb531ad65f3
res/res_pjsip_header_funcs.c
res/res_pjsip_send_to_voicemail.c [new file with mode: 0644]