From c4b34ef45d632388eb1dea1989c624ff9e8580b7 Mon Sep 17 00:00:00 2001
From: Mark Michelson <mmichelson@digium.com>
Date: Wed, 20 Aug 2008 15:38:47 +0000
Subject: [PATCH] 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
---
channels/chan_sip.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
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;
}
--
GitLab