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/1.4@139015 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 133ca3204270c9dad26d81696f46456f2352a12c..193d2a98aef8df6f45b73f4e7a6cd99684593285 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -4401,7 +4401,7 @@ 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;
        }