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Merged revisions 332027 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332027 | mnicholson | 2011-08-16 10:08:40 -0500 (Tue, 16 Aug 2011) | 9 lines Merged revisions 332026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332026 | mnicholson | 2011-08-16 10:06:31 -0500 (Tue, 16 Aug 2011) | 2 lines use DEFAULT_STORE_SIP_CAUSE to set the default value for the 'storesipcause' option AST-580 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -28111,7 +28111,7 @@ static int reload_config(enum channelreloadreason reason)
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global_shrinkcallerid = 1;
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authlimit = DEFAULT_AUTHLIMIT;
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authtimeout = DEFAULT_AUTHTIMEOUT;
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global_store_sip_cause = FALSE;
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global_store_sip_cause = DEFAULT_STORE_SIP_CAUSE;
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sip_cfg.matchexternaddrlocally = DEFAULT_MATCHEXTERNADDRLOCALLY;
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@@ -222,6 +222,7 @@
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#define DEFAULT_SDPSESSION "Asterisk PBX" /*!< Default SDP session name, (s=) header unless re-defined in sip.conf */
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#define DEFAULT_SDPOWNER "root" /*!< Default SDP username field in (o=) header unless re-defined in sip.conf */
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#define DEFAULT_ENGINE "asterisk" /*!< Default RTP engine to use for sessions */
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#define DEFAULT_STORE_SIP_CAUSE FALSE /*!< Store HASH(SIP_CAUSE,<channel name>) for channels by default */
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#endif
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/*@}*/
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