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Merged revisions 139015 via svnmerge from

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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: http://svn.digium.com/svn/asterisk/trunk@139016 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
mmichelson 2008-08-20 15:38:47 +00:00
parent a913c01fbc
commit f223c2f20e
1 changed files with 1 additions and 1 deletions
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2
channels/chan_sip.c
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@ -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;
}