[asterisk-commits] file: branch 1.4 r75621 - /branches/1.4/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: file
Date: Wed Jul 18 10:41:06 2007
New Revision: 75621

URL: http://svn.digium.com/view/asterisk?view=rev&rev=75621
Log:
(closes issue #10165)
Reported by: elandivar

It is possible for hold status to exist without call limits set, so we need to ensure update_call_counter is executed regardless.

Modified:
    branches/1.4/channels/chan_sip.c

Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=75621&r1=75620&r2=75621
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Wed Jul 18 10:41:06 2007
@@ -2943,7 +2943,7 @@
 	if (sip_debug_test_pvt(p) || option_debug > 2)
 		ast_verbose("Really destroying SIP dialog '%s' Method: %s\n", p->callid, sip_methods[p->method].text);
 
-	if (ast_test_flag(&p->flags[0], SIP_INC_COUNT)) {
+	if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
 		update_call_counter(p, DEC_CALL_LIMIT);
 		if (option_debug > 1)
 			ast_log(LOG_DEBUG, "This call did not properly clean up call limits. Call ID %s\n", p->callid);
@@ -3056,9 +3056,10 @@
 
 	if (option_debug > 2)
 		ast_log(LOG_DEBUG, "Updating call counter for %s call\n", outgoing ? "outgoing" : "incoming");
+
 	/* Test if we need to check call limits, in order to avoid 
 	   realtime lookups if we do not need it */
-	if (!ast_test_flag(&fup->flags[0], SIP_CALL_LIMIT))
+	if (!ast_test_flag(&fup->flags[0], SIP_CALL_LIMIT) && !ast_test_flag(&fup->flags[1], SIP_PAGE2_CALL_ONHOLD))
 		return 0;
 
 	ast_copy_string(name, fup->username, sizeof(name));