[asterisk-commits] oej: branch oej/pinefrog-1.4 r248405 - /team/oej/pinefrog-1.4/channels/

[SVN commits to the Asterisk project]

Author: oej
Date: Tue Feb 23 11:17:39 2010
New Revision: 248405

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=248405
Log:
What? No, I never wrote that code.

Modified:
    team/oej/pinefrog-1.4/channels/chan_sip.c

Modified: team/oej/pinefrog-1.4/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinefrog-1.4/channels/chan_sip.c?view=diff&rev=248405&r1=248404&r2=248405
==============================================================================
--- team/oej/pinefrog-1.4/channels/chan_sip.c (original)
+++ team/oej/pinefrog-1.4/channels/chan_sip.c Tue Feb 23 11:17:39 2010
@@ -13469,7 +13469,7 @@
 			"BridgedUniqueid: %s\r\n"
 			"RTPreporttype: %s\r\n"
 			"RTPrtcpstatus: %s\r\n"
-			"Duration: %ld\r\n"		/* used in cdr_manager */
+			"Duration: %u\r\n"		/* used in cdr_manager */
 			"PvtCallid: %s\r\n"		/* ??? Generic PVT identifier */
 			"RTPipaddress: %s\r\n"
 			"RTPmedia: %s\r\n"		/* Audio, video, text */
@@ -13589,7 +13589,7 @@
 		ast_rtp_stop(p->rtp);
 		sip_rtcp_report(p, p->rtp, "audio", TRUE);
 	}
-	if (p->vrtp) {
+	if (p->vrtp && ast_rtp_isactive(p->vrtp)) {
 		ast_rtp_stop(p->vrtp);
 		sip_rtcp_report(p, p->vrtp, "video", TRUE);
 	}