Import revision 13547 from branch 1.2 - reset global_rtautoclear at reload

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@13548 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 7214eea..d13fda4 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -391,7 +391,7 @@ static char default_musicclass[MAX_MUSICCLASS];             /*!< Global music on hold class
 static struct ast_codec_pref default_prefs;            /*!< Default codec prefs */
 
 /* Global settings only apply to the channel */
-static int global_rtautoclear = 120;
+static int global_rtautoclear;
 static int global_notifyringing;       /*!< Send notifications on ringing */
 static int srvlookup;                  /*!< SRV Lookup on or off. Default is off, RFC behavior is on */
 static int pedanticsipchecking;                /*!< Extra checking ?  Default off */
@@ -616,7 +616,6 @@ struct sip_auth {
 #define sipdebug_config                ast_test_flag(&global_flags_page2, SIP_PAGE2_DEBUG_CONFIG)
 #define sipdebug_console       ast_test_flag(&global_flags_page2, SIP_PAGE2_DEBUG_CONSOLE)
 
-
 /*! \brief sip_pvt: PVT structures are used for each SIP dialog, ie. a call, a registration, a subscribe  */
 static struct sip_pvt {
        ast_mutex_t lock;                       /*!< Dialog private lock */
@@ -12453,6 +12452,7 @@ static int reload_config(enum channelreloadreason reason)
        global_rtptimeout = 0;
        global_rtpholdtimeout = 0;
        global_rtpkeepalive = 0;
+       global_rtautoclear = 120;
        ast_set_flag(&global_flags_page2, SIP_PAGE2_RTUPDATE);
 
        /* Initialize some reasonable defaults at SIP reload (used both for channel and as default for peers and users */