[asterisk-commits] file: trunk r75624 - in /trunk: ./ channels/chan_sip.c

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Wed Jul 18 10:45:18 CDT 2007


Author: file
Date: Wed Jul 18 10:45:18 2007
New Revision: 75624

URL: http://svn.digium.com/view/asterisk?view=rev&rev=75624
Log:
Merged revisions 75623 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r75623 | file | 2007-07-18 12:44:02 -0300 (Wed, 18 Jul 2007) | 2 lines

Few more places that needs to check for onhold state.

........

Modified:
    trunk/   (props changed)
    trunk/channels/chan_sip.c

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=75624&r1=75623&r2=75624
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Wed Jul 18 10:45:18 2007
@@ -3683,7 +3683,7 @@
 	}
 
 	if (ast_test_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER)) {
-		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)) {
 			if (sipdebug)
 				ast_debug(1, "update_call_counter(%s) - decrement call limit counter on hangup\n", p->username);
 			update_call_counter(p, DEC_CALL_LIMIT);
@@ -3707,7 +3707,7 @@
 		ast_debug(1, "Hanging up zombie call. Be scared.\n");
 
 	sip_pvt_lock(p);
-	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)) {
 		if (sipdebug)
 			ast_debug(1, "update_call_counter(%s) - decrement call limit counter on hangup\n", p->username);
 		update_call_counter(p, DEC_CALL_LIMIT);
@@ -15011,7 +15011,7 @@
 		return 0;
 	}
 
-	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);
 
 	stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */




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