[asterisk-commits] branch oej/rtcp r11981 - in /team/oej/rtcp: ./ apps/ channels/ configs/

asterisk-commits at lists.digium.com asterisk-commits at lists.digium.com
Mon Mar 6 02:15:26 MST 2006


Author: oej
Date: Mon Mar  6 03:15:21 2006
New Revision: 11981

URL: http://svn.digium.com/view/asterisk?rev=11981&view=rev
Log:
Reset automerge

Modified:
    team/oej/rtcp/   (props changed)
    team/oej/rtcp/apps/app_mixmonitor.c
    team/oej/rtcp/channels/chan_sip.c
    team/oej/rtcp/channels/chan_zap.c
    team/oej/rtcp/configs/zapata.conf.sample

Propchange: team/oej/rtcp/
------------------------------------------------------------------------------
    automerge = yes

Propchange: team/oej/rtcp/
------------------------------------------------------------------------------
Binary property 'branch-1.2-blocked' - no diff available.

Propchange: team/oej/rtcp/
------------------------------------------------------------------------------
Binary property 'branch-1.2-merged' - no diff available.

Propchange: team/oej/rtcp/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Mon Mar  6 03:15:21 2006
@@ -1,1 +1,1 @@
-/trunk:1-11776
+/trunk:1-11979

Modified: team/oej/rtcp/apps/app_mixmonitor.c
URL: http://svn.digium.com/view/asterisk/team/oej/rtcp/apps/app_mixmonitor.c?rev=11981&r1=11980&r2=11981&view=diff
==============================================================================
--- team/oej/rtcp/apps/app_mixmonitor.c (original)
+++ team/oej/rtcp/apps/app_mixmonitor.c Mon Mar  6 03:15:21 2006
@@ -217,6 +217,17 @@
 	if (option_verbose > 1)
 		ast_verbose(VERBOSE_PREFIX_2 "Begin MixMonitor Recording %s\n", name);
 	
+	if (mixmonitor->post_process) {
+		char *p;
+
+		for (p = mixmonitor->post_process; *p ; p++) {
+			if (*p == '^' && *(p+1) == '{') {
+				*p = '$';
+			}
+		}
+		pbx_substitute_variables_helper(mixmonitor->chan, mixmonitor->post_process, post_process, sizeof(post_process) - 1);
+	}
+
 	while (1) {
 		struct ast_frame *next;
 		int write;
@@ -251,17 +262,6 @@
 		ast_mutex_unlock(&spy.lock);
 	}
 	
-	if (mixmonitor->post_process) {
-		char *p;
-
-		for (p = mixmonitor->post_process; *p ; p++) {
-			if (*p == '^' && *(p+1) == '{') {
-				*p = '$';
-			}
-		}
-		pbx_substitute_variables_helper(mixmonitor->chan, mixmonitor->post_process, post_process, sizeof(post_process) - 1);
-	}
-
 	stopmon(mixmonitor->chan, &spy);
 
 	if (option_verbose > 1)

Modified: team/oej/rtcp/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/rtcp/channels/chan_sip.c?rev=11981&r1=11980&r2=11981&view=diff
==============================================================================
--- team/oej/rtcp/channels/chan_sip.c (original)
+++ team/oej/rtcp/channels/chan_sip.c Mon Mar  6 03:15:21 2006
@@ -12339,6 +12339,7 @@
 	int auto_sip_domains = FALSE;
 	struct sockaddr_in old_bindaddr = bindaddr;
 	int registry_count = 0, peer_count = 0, user_count = 0;
+	struct ast_flags debugflag = {0};
 
 	cfg = ast_config_load(config);
 
@@ -12349,7 +12350,11 @@
 	}
 	
 	/* Clear all flags before setting default values */
+	/* Preserve debugging settings for console */
+	ast_copy_flags((&debugflag), (&global_flags_page2), SIP_PAGE2_DEBUG_CONSOLE);
 	ast_clear_flag(&global_flags, AST_FLAGS_ALL);
+	ast_clear_flag(&global_flags_page2, AST_FLAGS_ALL);
+	ast_copy_flags((&global_flags_page2), (&debugflag), SIP_PAGE2_DEBUG_CONSOLE);
 
 	/* Reset IP addresses  */
 	memset(&bindaddr, 0, sizeof(bindaddr));

Modified: team/oej/rtcp/channels/chan_zap.c
URL: http://svn.digium.com/view/asterisk/team/oej/rtcp/channels/chan_zap.c?rev=11981&r1=11980&r2=11981&view=diff
==============================================================================
--- team/oej/rtcp/channels/chan_zap.c (original)
+++ team/oej/rtcp/channels/chan_zap.c Mon Mar  6 03:15:21 2006
@@ -4299,24 +4299,24 @@
 			}
 			break;
 		case ZT_EVENT_POLARITY:
-                        /*
-                         * If we get a Polarity Switch event, check to see
-                         * if we should change the polarity state and
-                         * mark the channel as UP or if this is an indication
-                         * of remote end disconnect.
-                         */
-                        if (p->polarity == POLARITY_IDLE) {
-                                p->polarity = POLARITY_REV;
-                                if (p->answeronpolarityswitch &&
-                                    ((ast->_state == AST_STATE_DIALING) ||
-                                     (ast->_state == AST_STATE_RINGING))) {
-                                        ast_log(LOG_DEBUG, "Answering on polarity switch!\n");
-                                        ast_setstate(p->owner, AST_STATE_UP);
+			/*
+			 * If we get a Polarity Switch event, check to see
+			 * if we should change the polarity state and
+			 * mark the channel as UP or if this is an indication
+			 * of remote end disconnect.
+			 */
+			if (p->polarity == POLARITY_IDLE) {
+				p->polarity = POLARITY_REV;
+				if (p->answeronpolarityswitch &&
+				    ((ast->_state == AST_STATE_DIALING) ||
+					 (ast->_state == AST_STATE_RINGING))) {
+					ast_log(LOG_DEBUG, "Answering on polarity switch!\n");
+					ast_setstate(p->owner, AST_STATE_UP);
 					if (p->hanguponpolarityswitch) {
 						gettimeofday(&p->polaritydelaytv, NULL);
 					}
-                                } else
-                                        ast_log(LOG_DEBUG, "Ignore switch to REVERSED Polarity on channel %d, state %d\n", p->channel, ast->_state);
+				} else
+					ast_log(LOG_DEBUG, "Ignore switch to REVERSED Polarity on channel %d, state %d\n", p->channel, ast->_state);
 			} 
 			/* Removed else statement from here as it was preventing hangups from ever happening*/
 			/* Added AST_STATE_RING in if statement below to deal with calling party hangups that take place when ringing */

Modified: team/oej/rtcp/configs/zapata.conf.sample
URL: http://svn.digium.com/view/asterisk/team/oej/rtcp/configs/zapata.conf.sample?rev=11981&r1=11980&r2=11981&view=diff
==============================================================================
--- team/oej/rtcp/configs/zapata.conf.sample (original)
+++ team/oej/rtcp/configs/zapata.conf.sample Mon Mar  6 03:15:21 2006
@@ -444,7 +444,7 @@
 ;callprogress=yes
 ;progzone=us
 ;
-; FXO (FXS signalled) devices must have a timeout to determine whe there was a
+; FXO (FXS signalled) devices must have a timeout to determine if there was a
 ; hangup before the line was answered.  This value can be tweaked to shorten
 ; how long it takes before Zap considers a non-ringing line to have hungup.
 ;



More information about the asterisk-commits mailing list