From fe1294fbd8574a1e828a97c666ed4bd6ccdad37d Mon Sep 17 00:00:00 2001 From: oej Date: Mon, 7 Sep 2009 16:16:58 +0000 Subject: [PATCH] Doxygen updates git-svn-id: http://svn.digium.com/svn/asterisk/trunk@216806 f38db490-d61c-443f-a65b-d21fe96a405b --- channels/chan_sip.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 1dea03cf4..d1aa7178a 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -1080,7 +1080,8 @@ static const char *sip_reason_code_to_str(enum AST_REDIRECTING_REASON code) * 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 @@ static const char *sip_reason_code_to_str(enum AST_REDIRECTING_REASON code) #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 @@ struct sip_settings { 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_prematuremediafilter; /*!< Enable/disable premature frames in 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