[asterisk-commits] mmichelson: branch 1.6.1 r139018 - in /branches/1.6.1: ./ channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: mmichelson
Date: Wed Aug 20 10:39:39 2008
New Revision: 139018

URL: http://svn.digium.com/view/asterisk?view=rev&rev=139018
Log:
Merged revisions 139016 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
r139016 | mmichelson | 2008-08-20 10:38:47 -0500 (Wed, 20 Aug 2008) | 14 lines

Merged revisions 139015 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r139015 | mmichelson | 2008-08-20 10:37:56 -0500 (Wed, 20 Aug 2008) | 6 lines

sip_read should properly handle a NULL return from sip_rtp_read.

(closes issue #13257)
Reported by: travishein


........

................

Modified:
    branches/1.6.1/   (props changed)
    branches/1.6.1/channels/chan_sip.c

Propchange: branches/1.6.1/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.

Modified: branches/1.6.1/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.6.1/channels/chan_sip.c?view=diff&rev=139018&r1=139017&r2=139018
==============================================================================
--- branches/1.6.1/channels/chan_sip.c (original)
+++ branches/1.6.1/channels/chan_sip.c Wed Aug 20 10:39:39 2008
@@ -5912,7 +5912,7 @@
 	}
 
 	/* Only allow audio through if they sent progress with SDP, or if the channel is actually answered */
-	if (fr->frametype == AST_FRAME_VOICE && p->invitestate != INV_EARLY_MEDIA && ast->_state != AST_STATE_UP) {
+	if (fr && fr->frametype == AST_FRAME_VOICE && p->invitestate != INV_EARLY_MEDIA && ast->_state != AST_STATE_UP) {
 		fr = &ast_null_frame;
 	}