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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@139016 f38db490-d61c-443f-a65b-d21fe96a405b
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@ -5912,7 +5912,7 @@ static struct ast_frame *sip_read(struct ast_channel *ast)
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}
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/* Only allow audio through if they sent progress with SDP, or if the channel is actually answered */
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if (fr->frametype == AST_FRAME_VOICE && p->invitestate != INV_EARLY_MEDIA && ast->_state != AST_STATE_UP) {
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if (fr && fr->frametype == AST_FRAME_VOICE && p->invitestate != INV_EARLY_MEDIA && ast->_state != AST_STATE_UP) {
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fr = &ast_null_frame;
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}
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