Merged revisions 192634 via svnmerge from

https://origsvn.digium.com/svn/asterisk/trunk

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  r192634 | file | 2009-05-06 10:34:35 -0300 (Wed, 06 May 2009) | 14 lines
  
  Merged revisions 192633 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r192633 | file | 2009-05-06 10:30:51 -0300 (Wed, 06 May 2009) | 7 lines
    
    Update some old logic to stop both begin and end DTMF frames from reaching the core if rfc2833 is not enabled.
    
    (closes issue #15036)
    Reported by: dimas
    Patches:
          v1-15036.patch uploaded by dimas (license 88)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@192636 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -6077,9 +6077,11 @@ static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p
 		f = &ast_null_frame;
 	}
 	/* Don't forward RFC2833 if we're not supposed to */
-	if (f && (f->frametype == AST_FRAME_DTMF) &&
-	    (ast_test_flag(&p->flags[0], SIP_DTMF) != SIP_DTMF_RFC2833))
+	if (f && (f->frametype == AST_FRAME_DTMF_BEGIN || f->frametype == AST_FRAME_DTMF_END) &&
+	    (ast_test_flag(&p->flags[0], SIP_DTMF) != SIP_DTMF_RFC2833)) {
+		ast_debug(1, "Ignoring DTMF (%c) RTP frame because dtmfmode is not RFC2833\n", f->subclass);
 		return &ast_null_frame;
+	}
 
 	/* We already hold the channel lock */
 	if (!p->owner || (f && f->frametype != AST_FRAME_VOICE))