From 3c0fbfd326d22ef6751ea75fc753becd5140312d Mon Sep 17 00:00:00 2001 From: Mark Michelson Date: Wed, 20 Aug 2008 15:39:39 +0000 Subject: [PATCH] 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 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@139018 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; }