[asterisk-commits] oej: trunk r216806 - /trunk/channels/chan_sip.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Sep 7 11:17:00 CDT 2009
Author: oej
Date: Mon Sep 7 11:16:58 2009
New Revision: 216806
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=216806
Log:
Doxygen updates
Modified:
trunk/channels/chan_sip.c
Modified: trunk/channels/chan_sip.c
URL: http://svn.asterisk.org/svn-view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=216806&r1=216805&r2=216806
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Mon Sep 7 11:16:58 2009
@@ -1080,7 +1080,8 @@
* in principle, use a different "default" port number, but
* we do not support this feature at the moment.
* You can run Asterisk with SIP on a different port with a configuration
- * option. If you change this value, the signalling will be incorrect.
+ * option. If you change this value in the source code, the signalling will be incorrect.
+ *
*/
/*! \name DefaultValues Default values, set and reset in reload_config before reading configuration
@@ -1099,7 +1100,7 @@
#define DEFAULT_NOTIFYMIME "application/simple-message-summary"
#define DEFAULT_ALLOWGUEST TRUE
#define DEFAULT_RTPKEEPALIVE 0 /*!< Default RTPkeepalive setting */
-#define DEFAULT_CALLCOUNTER FALSE
+#define DEFAULT_CALLCOUNTER FALSE /*!< Do not enable call counters by default */
#define DEFAULT_SRVLOOKUP TRUE /*!< Recommended setting is ON */
#define DEFAULT_COMPACTHEADERS FALSE /*!< Send compact (one-character) SIP headers. Default off */
#define DEFAULT_TOS_SIP 0 /*!< Call signalling packets should be marked as DSCP CS3, but the default is 0 to be compatible with previous versions. */
@@ -1196,7 +1197,8 @@
char default_subscribecontext[AST_MAX_CONTEXT];
};
-static struct sip_settings sip_cfg;
+static struct sip_settings sip_cfg; /*!< SIP configuration data.
+ \note in the future we could have multiple of these (per domain, per device group etc) */
static int global_match_auth_username; /*!< Match auth username if available instead of From: Default off. */
@@ -1205,7 +1207,7 @@
static int global_rtptimeout; /*!< Time out call if no RTP */
static int global_rtpholdtimeout; /*!< Time out call if no RTP during hold */
static int global_rtpkeepalive; /*!< Send RTP keepalives */
-static int global_reg_timeout;
+static int global_reg_timeout; /*!< Global time between attempts for outbound registrations */
static int global_regattempts_max; /*!< Registration attempts before giving up */
static int global_callcounter; /*!< Enable call counters for all devices. This is currently enabled by setting the peer
call-limit to UINT_MAX. When we remove the call-limit from the code, we can make it
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