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Import revision 13547 from branch 1.2 - reset global_rtautoclear at reload

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@13548 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson
2006-03-19 09:32:36 +00:00
parent 7296dd030e
commit 8bb397b439
1 changed files with 2 additions and 2 deletions
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4
channels/chan_sip.c
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@@ -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 */