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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332028 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -28111,7 +28111,7 @@ static int reload_config(enum channelreloadreason reason)
 	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;
 

channels/sip/include/sip.h

@@ -222,6 +222,7 @@
 #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
 /*@}*/