Merged revisions 336791 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r336791 | twilson | 2011-09-19 17:07:58 -0500 (Mon, 19 Sep 2011) | 2 lines
  
  Don't interfere with T.38 reinvites

  This is an update to the fix for ASTERISK-18340 and ASTERISK-17725
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336792 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -20115,7 +20115,11 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
 						"Channel: %s\r\nChanneltype: %s\r\nUniqueid: %s\r\nSIPcallid: %s\r\nSIPfullcontact: %s\r\nPeername: %s\r\n",
 						p->owner->name, "SIP", p->owner->uniqueid, p->callid, p->fullcontact, p->peername);
 			} else {	/* RE-invite */
-				ast_queue_control(p->owner, AST_CONTROL_UPDATE_RTP_PEER);
+				if (p->t38.state == T38_DISABLED || p->t38.state == T38_REJECTED) {
+					ast_queue_control(p->owner, AST_CONTROL_UPDATE_RTP_PEER);
+				} else {
+					ast_queue_frame(p->owner, &ast_null_frame);
+				}
 			}
 		} else {
 			 /* It's possible we're getting an 200 OK after we've tried to disconnect