Correct some scenarios where CALL_LIMIT could not be getting adjusted properly allow...

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8432 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index dc0df0d..e05793c 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -567,6 +567,8 @@ struct sip_auth {
 #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 @@ static int update_call_counter(struct sip_pvt *fup, int event)
                /* 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 @@ static int update_call_counter(struct sip_pvt *fup, int event)
                                }
                        }
                        (*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);
                        }