Index: asterisk/channels/chan_sip.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.836
diff -u -r1.836 chan_sip.c
--- asterisk/channels/chan_sip.c	2 Sep 2005 19:24:32 -0000	1.836
+++ asterisk/channels/chan_sip.c	6 Sep 2005 20:40:18 -0000
@@ -2143,7 +2143,9 @@
 			} else {
 				*inuse = 0;
 			}
-			ast_log(LOG_ERROR, "Call %s %s '%s' removed from call limit %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit);
+			if (option_debug > 1 || sipdebug) {
+				ast_log(LOG_DEBUG, "Call %s %s '%s' removed from call limit %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit);
+			}
 			break;
 		case INC_CALL_LIMIT:
 			if (*call_limit > 0 ) {
@@ -2161,7 +2163,9 @@
 				}
 			}
 			(*inuse)++;
-			ast_log(LOG_DEBUG, "Call %s %s '%s' is %d out of %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *inuse, *call_limit);
+			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);
+			}
 			break;
 		default:
 			ast_log(LOG_ERROR, "update_call_counter(%s, %d) called with no event!\n", name, event);