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authorMark Michelson <mmichelson@digium.com>2008-08-20 15:38:47 +0000
committerMark Michelson <mmichelson@digium.com>2008-08-20 15:38:47 +0000
commitc4b34ef45d632388eb1dea1989c624ff9e8580b7 (patch)
treed0543a561df0ba735db3249acaf87ceff0fa360d /channels
parent78c68e8015f5ab713eae26af4c9520e53551fc12 (diff)
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
Diffstat (limited to 'channels')
-rw-r--r--channels/chan_sip.c2
1 files changed, 1 insertions, 1 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index b6f169f04..01cca0542 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;
}