[asterisk-commits] trunk r13548 - in /trunk: ./ channels/chan_sip.c

[asterisk-commits at lists.digium.com]

Author: oej
Date: Sun Mar 19 03:32:36 2006
New Revision: 13548

URL: http://svn.digium.com/view/asterisk?rev=13548&view=rev
Log:
Import revision 13547 from branch 1.2 - reset global_rtautoclear at reload

Modified:
    trunk/   (props changed)
    trunk/channels/chan_sip.c

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.2-merged' - no diff available.

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=13548&r1=13547&r2=13548&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Sun Mar 19 03:32:36 2006
@@ -391,7 +391,7 @@
 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 */
@@ -615,7 +615,6 @@
 #define sipdebug		ast_test_flag(&global_flags_page2, SIP_PAGE2_DEBUG)
 #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 {
@@ -12453,6 +12452,7 @@
 	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 */