[svn-commits] oej: branch oej/codename-pineapple r48203 - in /team/oej/codename-pineapple/c...

[svn-commits at lists.digium.com]

Author: oej
Date: Sat Dec  2 06:53:43 2006
New Revision: 48203

URL: http://svn.digium.com/view/asterisk?view=rev&rev=48203
Log:
Bug fixing.

Modified:
    team/oej/codename-pineapple/channels/chan_sip3.c
    team/oej/codename-pineapple/channels/sip3/sip3_monitor.c
    team/oej/codename-pineapple/channels/sip3/sip3_sdprtp.c

Modified: team/oej/codename-pineapple/channels/chan_sip3.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/chan_sip3.c?view=diff&rev=48203&r1=48202&r2=48203
==============================================================================
--- team/oej/codename-pineapple/channels/chan_sip3.c (original)
+++ team/oej/codename-pineapple/channels/chan_sip3.c Sat Dec  2 06:53:43 2006
@@ -991,9 +991,14 @@
 	else
 		dialog->noncodeccapability &= ~AST_RTP_DTMF;
 	ast_string_field_set(dialog, context, device->context);
-	dialog->rtptimeout = device->rtptimeout;
-	dialog->rtpholdtimeout = device->rtpholdtimeout;
-	dialog->rtpkeepalive = device->rtpkeepalive;
+	ast_rtp_set_rtptimeout(dialog->rtp, device->rtptimeout);
+	ast_rtp_set_rtpholdtimeout(dialog->rtp, device->rtpholdtimeout);
+	ast_rtp_set_rtpkeepalive(dialog->rtp, device->rtpkeepalive);
+	if (dialog->vrtp) {
+		ast_rtp_set_rtptimeout(dialog->vrtp, device->rtptimeout);
+		ast_rtp_set_rtpholdtimeout(dialog->vrtp, device->rtpholdtimeout);
+		ast_rtp_set_rtpkeepalive(dialog->vrtp, device->rtpkeepalive);
+	}
 	if (device->call_limit)
 		ast_set_flag(&dialog->flags[0], SIP_CALL_LIMIT);
 	dialog->maxcallbitrate = device->maxcallbitrate;
@@ -5245,7 +5250,7 @@
 		/* we need to check the tags and the branch. If they're different, this is
 	   	  in fact a forked call through a SIP proxy somewhere. */
 		transmit_final_response(p, "482 Loop Detected", req, XMIT_RELIABLE);
-		sip_sched_destroy(p, DEFAULT_TRANS_TIMEOUT);
+		sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 		return 0;
 	}
 	

Modified: team/oej/codename-pineapple/channels/sip3/sip3_monitor.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/sip3/sip3_monitor.c?view=diff&rev=48203&r1=48202&r2=48203
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3_monitor.c (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3_monitor.c Sat Dec  2 06:53:43 2006
@@ -157,7 +157,7 @@
 /*! \brief Check RTP timeouts and send RTP keepalives 
 	\note We're only sending audio keepalives yet.	
 */
-static void check_rtp_timeout(struct sip_pvt *dialog, time_t t)
+static void check_rtp_timeout(struct sip_dialog *dialog, time_t t)
 {
 	/* If we have no RTP or no active owner, no need to check timers */
 	if (!dialog->rtp || !dialog->owner)
@@ -194,11 +194,11 @@
 		if (sin.sin_addr.s_addr || (ast_rtp_get_rtpholdtimeout(dialog->rtp) &&
 		     (t > dialog->lastrtprx + ast_rtp_get_rtpholdtimeout(dialog->rtp)))) {
 			/* Needs a hangup */
-			if (dialog->rtptimeout) {
+			if (ast_rtp_get_rtptimeout(dialog->rtp)) {
 				while (dialog->owner && ast_channel_trylock(dialog->owner)) {
-					sip_pvt_unlock(dialog);
+					dialog_lock(dialog, FALSE);
 					usleep(1);
-					sip_pvt_lock(dialog);
+					dialog_lock(dialog, TRUE);
 				}
 				if (!(ast_rtp_get_bridged(dialog->rtp))) {
 					ast_log(LOG_NOTICE, "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",

Modified: team/oej/codename-pineapple/channels/sip3/sip3_sdprtp.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/sip3/sip3_sdprtp.c?view=diff&rev=48203&r1=48202&r2=48203
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3_sdprtp.c (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3_sdprtp.c Sat Dec  2 06:53:43 2006
@@ -378,7 +378,7 @@
 			if (debug)
 				ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
 		} else {
-			if (utptlportno > 0) {
+			if (udptlportno > 0) {
 				if (debug)
 					ast_log(LOG_DEBUG, "Got T.38 re-invite without audio. Keeping RTP active during T.38 session. Call-Id %s\n", p->callid);
 			} else