[asterisk-commits] trunk - r8432 /trunk/channels/chan_sip.c

[asterisk-commits at lists.digium.com]

Author: bweschke
Date: Sun Jan 22 09:08:51 2006
New Revision: 8432

URL: http://svn.digium.com/view/asterisk?rev=8432&view=rev
Log:
 Correct some scenarios where CALL_LIMIT could not be getting adjusted properly allowing chan_sip to send calls when it really shouldn't. Bug #6111

Modified:
    trunk/channels/chan_sip.c

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=8432&r1=8431&r2=8432&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Sun Jan 22 09:08:51 2006
@@ -567,6 +567,8 @@
 #define SIP_CALL_LIMIT		(1 << 29)
 /* Remote Party-ID Support */
 #define SIP_SENDRPID		(1 << 30)
+/* Did this connection increment the counter of in-use calls? */
+#define SIP_INC_COUNT (1 << 31)
 
 #define SIP_FLAGS_TO_COPY \
 	(SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_SENDRPID | SIP_DTMF | SIP_REINVITE | \
@@ -2229,7 +2231,8 @@
 		/* incoming and outgoing affects the inUse counter */
 		case DEC_CALL_LIMIT:
 			if ( *inuse > 0 ) {
-				(*inuse)--;
+			         if (ast_test_flag(fup,SIP_INC_COUNT))
+				         (*inuse)--;
 			} else {
 				*inuse = 0;
 			}
@@ -2249,6 +2252,7 @@
 				}
 			}
 			(*inuse)++;
+	                ast_set_flag(fup,SIP_INC_COUNT);
 			if (option_debug > 1 || sipdebug) {
 				ast_log(LOG_DEBUG, "Call %s %s '%s' is %d out of %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *inuse, *call_limit);
 			}