[asterisk-commits] mmichelson: branch 1.4 r139015 - /branches/1.4/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: mmichelson
Date: Wed Aug 20 10:37:56 2008
New Revision: 139015

URL: http://svn.digium.com/view/asterisk?view=rev&rev=139015
Log:
sip_read should properly handle a NULL return from sip_rtp_read.

(closes issue #13257)
Reported by: travishein


Modified:
    branches/1.4/channels/chan_sip.c

Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=139015&r1=139014&r2=139015
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Wed Aug 20 10:37:56 2008
@@ -4401,7 +4401,7 @@
 	}
 
 	/* 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;
 	}