diff options
author | Joshua Colp <jcolp@digium.com> | 2014-11-03 14:45:01 +0000 |
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committer | Joshua Colp <jcolp@digium.com> | 2014-11-03 14:45:01 +0000 |
commit | ac091d41844a9a4a0f7d539164bcd154351b6da7 (patch) | |
tree | 84ec4d1350b4e6d1d1498c4ceabd2b5484f3947d /include/asterisk/res_pjsip.h | |
parent | 285be15aaf0469055d3392ecd73eb24395e49059 (diff) |
chan_pjsip: Add support for passing hold and unhold requests through.
This change adds an option, moh_passthrough, that when enabled will pass
hold and unhold requests through using a SIP re-invite. When placing on
hold a re-invite with sendonly will be sent and when taking off hold a
re-invite with sendrecv will be sent. This allows remote servers to handle
the musiconhold instead of the local Asterisk instance being responsible.
Review: https://reviewboard.asterisk.org/r/4103/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427112 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'include/asterisk/res_pjsip.h')
-rw-r--r-- | include/asterisk/res_pjsip.h | 2 |
1 files changed, 2 insertions, 0 deletions
diff --git a/include/asterisk/res_pjsip.h b/include/asterisk/res_pjsip.h index 1fe0b040e..1c21c1ee4 100644 --- a/include/asterisk/res_pjsip.h +++ b/include/asterisk/res_pjsip.h @@ -609,6 +609,8 @@ struct ast_sip_endpoint { struct ast_variable *channel_vars; /*! Whether to place a 'user=phone' parameter into the request URI if user is a number */ unsigned int usereqphone; + /*! Whether to pass through hold and unhold using re-invites with recvonly and sendrecv */ + unsigned int moh_passthrough; }; /*! |