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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@216806 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
oej 2009-09-07 16:16:58 +00:00
parent c8e7d6a547
commit fe1294fbd8
1 changed files with 6 additions and 4 deletions
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10
channels/chan_sip.c
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@ -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