[svn-commits] branch oej/siptransfer r21078 - in /team/oej/siptransfer: ./ apps/ channels/

[svn-commits at lists.digium.com]

Author: oej
Date: Tue Apr 18 02:30:59 2006
New Revision: 21078

URL: http://svn.digium.com/view/asterisk?rev=21078&view=rev
Log:
Reset, resolve, go

Modified:
    team/oej/siptransfer/   (props changed)
    team/oej/siptransfer/apps/app_random.c
    team/oej/siptransfer/channels/chan_sip.c

Propchange: team/oej/siptransfer/
------------------------------------------------------------------------------
    automerge = http://edvina.net/training/

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

Propchange: team/oej/siptransfer/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Tue Apr 18 02:30:59 2006
@@ -1,1 +1,1 @@
-/trunk:1-21025
+/trunk:1-21077

Modified: team/oej/siptransfer/apps/app_random.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/apps/app_random.c?rev=21078&r1=21077&r2=21078&view=diff
==============================================================================
--- team/oej/siptransfer/apps/app_random.c (original)
+++ team/oej/siptransfer/apps/app_random.c Tue Apr 18 02:30:59 2006
@@ -82,10 +82,10 @@
 
 	if (!deprecated) {
 		deprecated = 1;
-		ast_log(LOG_WARNING, "Random is deprecated in Asterisk 1.3 or later.  Replace with GotoIf($[${RAND(0,99)} + %d >= 100]?%s)\n", probint, s);
+		ast_log(LOG_WARNING, "Random is deprecated in Asterisk 1.4.  Replace with GotoIf($[${RAND(0,99)} + %d >= 100]?%s)\n", probint, s);
 	}
 
-	if ((ast_random() % 100) + probint > 100) {
+	if ((ast_random() % 100) + probint >= 100) {
 		res = ast_parseable_goto(chan, s);
 		if (option_verbose > 2)
 			ast_verbose( VERBOSE_PREFIX_3 "Random branches to (%s,%s,%d)\n",

Modified: team/oej/siptransfer/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/channels/chan_sip.c?rev=21078&r1=21077&r2=21078&view=diff
==============================================================================
--- team/oej/siptransfer/channels/chan_sip.c (original)
+++ team/oej/siptransfer/channels/chan_sip.c Tue Apr 18 02:30:59 2006
@@ -753,6 +753,7 @@
 		AST_STRING_FIELD(exten);	/*!< Extension where to start */
 		AST_STRING_FIELD(context);	/*!< Context for this call */
 		AST_STRING_FIELD(subscribecontext); /*!< Subscribecontext */
+		AST_STRING_FIELD(subscribeuri); /*!< Subscribecontext */
 		AST_STRING_FIELD(fromdomain);	/*!< Domain to show in the from field */
 		AST_STRING_FIELD(fromuser);	/*!< User to show in the user field */
 		AST_STRING_FIELD(fromname);	/*!< Name to show in the user field */
@@ -2812,7 +2813,7 @@
 		ast_setstate(ast, AST_STATE_UP);
 		if (option_debug)
 			ast_log(LOG_DEBUG, "SIP answering channel: %s\n", ast->name);
-		res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_RELIABLE);
+		res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
 	}
 	ast_mutex_unlock(&p->lock);
 	return res;
@@ -11483,7 +11484,7 @@
 		case AST_STATE_UP:
 			if (option_debug > 1)
 				ast_log(LOG_DEBUG, "%s: New call is UP.... \n", c->name);
-			transmit_response_with_sdp(p, "200 OK", req, 1);
+			transmit_response_with_sdp(p, "200 OK", req, XMIT_CRITICAL);
 
 			break;
 		default: