From: Mark Michelson <mmichelson@digium.com>
Date: Wed, 20 Aug 2008 15:38:47 +0000 (+0000)
Subject: Merged revisions 139015 via svnmerge from
X-Git-Tag: 1.6.2.0-beta1~1389
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=c4b34ef45d632388eb1dea1989c624ff9e8580b7;p=thirdparty%2Fasterisk.git

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


........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@139016 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index b6f169f04a..01cca05429 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -5912,7 +5912,7 @@ static struct ast_frame *sip_read(struct ast_channel *ast)
 	}
 
 	/* 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;
 	}