[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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