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

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@139016 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index b6f169f04afa80b552ec819d9151cf7de4fae29a..01cca054292440a0cf5905049f579a412540d150 100644 (file)
--- 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;
        }