* 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
#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. */
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. */
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