[asterisk-commits] oej: trunk r179675 - /trunk/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: oej
Date: Tue Mar  3 09:13:42 2009
New Revision: 179675

URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=179675
Log:
Please prefix default values with DEFAULT

Modified:
    trunk/channels/chan_sip.c

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/svn-view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=179675&r1=179674&r2=179675
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue Mar  3 09:13:42 2009
@@ -535,8 +535,8 @@
 
 #define DEFAULT_RETRANS              1000             /*!< How frequently to retransmit Default: 2 * 500 ms in RFC 3261 */
 #define MAX_RETRANS                  6                /*!< Try only 6 times for retransmissions, a total of 7 transmissions */
-#define SIP_TIMER_T1                 500              /*!< SIP timer T1 (according to RFC 3261) */
-#define SIP_TRANS_TIMEOUT            64 * SIP_TIMER_T1/*!< SIP request timeout (rfc 3261) 64*T1 
+#define DEFAULT_TIMER_T1                 500              /*!< SIP timer T1 (according to RFC 3261) */
+#define SIP_TRANS_TIMEOUT            64 * DEFAULT_TIMER_T1 /*!< SIP request timeout (rfc 3261) 64*T1 
                                                       \todo Use known T1 for timeout (peerpoke)
                                                       */
 #define DEFAULT_TRANS_TIMEOUT        -1               /*!< Use default SIP transaction timeout */
@@ -23187,8 +23187,8 @@
 	global_relaxdtmf = FALSE;
 	sip_cfg.callevents = DEFAULT_CALLEVENTS;
 	global_authfailureevents = FALSE;
-	global_t1 = SIP_TIMER_T1;
-	global_timer_b = 64 * SIP_TIMER_T1;
+	global_t1 = DEFAULT_TIMER_T1;
+	global_timer_b = 64 * DEFAULT_TIMER_T1;
 	global_t1min = DEFAULT_T1MIN;
 	global_qualifyfreq = DEFAULT_QUALIFYFREQ;