diff options
| author | Kevin Harwell <kharwell@digium.com> | 2014-02-25 17:47:06 +0000 |
|---|---|---|
| committer | Kevin Harwell <kharwell@digium.com> | 2014-02-25 17:47:06 +0000 |
| commit | eee4313fe8c1563e64d557c46d8e070275d07e99 (patch) | |
| tree | bcb3c69e1fa59c96cd584b251194f57d6474e6e2 /res/res_pjsip_exten_state.c | |
| parent | 23b142d5c8ca6c9d04e2a71be9dfbc1ff4c1527e (diff) | |
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
Diffstat (limited to 'res/res_pjsip_exten_state.c')
0 files changed, 0 insertions, 0 deletions