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authorJoshua Colp <jcolp@digium.com>2014-11-03 14:45:01 +0000
committerJoshua Colp <jcolp@digium.com>2014-11-03 14:45:01 +0000
commitac091d41844a9a4a0f7d539164bcd154351b6da7 (patch)
tree84ec4d1350b4e6d1d1498c4ceabd2b5484f3947d /include/asterisk/res_pjsip.h
parent285be15aaf0469055d3392ecd73eb24395e49059 (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.h2
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;
};
/*!