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authormmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b>2008-08-20 15:37:56 +0000
committermmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b>2008-08-20 15:37:56 +0000
commitdd0c9f3ad6b00177a24f38d79cc1aea1e27ab22b (patch)
tree0e24d800fab2cf1586594dde6a8faa67b00af7b2
parentae6c046916aad221cb29980ddf4a57704dfe26f4 (diff)
sip_read should properly handle a NULL return from sip_rtp_read.
(closes issue #13257) Reported by: travishein git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@139015 f38db490-d61c-443f-a65b-d21fe96a405b
-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 133ca3204..193d2a98a 100644
--- 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;
}