Merged revisions 139015 via svnmerge from
authorMark Michelson <mmichelson@digium.com>
Wed, 20 Aug 2008 15:38:47 +0000 (15:38 +0000)
committerMark Michelson <mmichelson@digium.com>
Wed, 20 Aug 2008 15:38:47 +0000 (15:38 +0000)
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

channels/chan_sip.c

index b6f169f..01cca05 100644 (file)
@@ -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;
        }