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authorOlle Johansson <oej@edvina.net>2007-07-09 08:27:37 +0000
committerOlle Johansson <oej@edvina.net>2007-07-09 08:27:37 +0000
commita1b9cbcd31d3fdc917ca60304913dec4232842b6 (patch)
tree023b2775a78d84d074ca62e3f349240deaf0c4d9 /include
parentba857cc8a9984225a842137c0674df6089862bb0 (diff)
Implementation of a feature that will disable "missed calls" counters on SIP phones.
If the call is answered by another phone, other phones won't display the call as "missed". You can also add an option to the dial command so that you can have a "followme" scenario and not count the calls as "missed" when you cancel the call. Thanks to Ramon and Frank for feedback on this feature. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@74024 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'include')
-rw-r--r--include/asterisk/channel.h3
1 files changed, 3 insertions, 0 deletions
diff --git a/include/asterisk/channel.h b/include/asterisk/channel.h
index 8c0b6a51a..139a8b10e 100644
--- a/include/asterisk/channel.h
+++ b/include/asterisk/channel.h
@@ -539,6 +539,9 @@ enum {
/*! This is set to tell the channel not to generate DTMF begin frames, and
* to instead only generate END frames. */
AST_FLAG_END_DTMF_ONLY = (1 << 14),
+ /*! Flag to show channels that this call is hangup due to the fact that the call
+ was indeed anwered, but in another channel */
+ AST_FLAG_ANSWERED_ELSEWHERE = (1 << 15),
};
/*! \brief ast_bridge_config flags */