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
global_shrinkcallerid = 1;
authlimit = DEFAULT_AUTHLIMIT;
authtimeout = DEFAULT_AUTHTIMEOUT;
- global_store_sip_cause = FALSE;
+ global_store_sip_cause = DEFAULT_STORE_SIP_CAUSE;
sip_cfg.matchexternaddrlocally = DEFAULT_MATCHEXTERNADDRLOCALLY;
#define DEFAULT_SDPSESSION "Asterisk PBX" /*!< Default SDP session name, (s=) header unless re-defined in sip.conf */
#define DEFAULT_SDPOWNER "root" /*!< Default SDP username field in (o=) header unless re-defined in sip.conf */
#define DEFAULT_ENGINE "asterisk" /*!< Default RTP engine to use for sessions */
+#define DEFAULT_STORE_SIP_CAUSE FALSE /*!< Store HASH(SIP_CAUSE,<channel name>) for channels by default */
#endif
/*@}*/