Merged revisions 160481 via svnmerge from

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

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  r160481 | tilghman | 2008-12-03 08:11:53 -0600 (Wed, 03 Dec 2008) | 14 lines
  
  Merged revisions 160480 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r160480 | tilghman | 2008-12-03 08:09:35 -0600 (Wed, 03 Dec 2008) | 7 lines
    
    Jon Bonilla (Manwe) pointed out on the -dev list:
    "I guess that having only ip-phones in mind is not a good approach. Since it is
    possible to have a sip proxy connected to asterisk we could receive a 407
    (unauthorized) or 483 (too many hops) as response and dialog ending would not be
    a good behavior."
    So modified.
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160482 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -15451,13 +15451,12 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
 
 	/* RFC 3261 Section 15 specifies that if we receive a 408 or 481
 	 * in response to a BYE, then we should end the current dialog
-	 * and session. There is no mention in the spec of other 4XX responses,
-	 * but it is known that at least one phone manufacturer potentially
-	 * will send a 404 in response to a BYE, so we'll be liberal in what
-	 * we accept and end the dialog and session if we receive any 4XX 
-	 * response to a BYE.
+	 * and session.  It is known that at least one phone manufacturer
+	 * potentially will send a 404 in response to a BYE, so we'll be
+	 * liberal in what we accept and end the dialog and session if we
+	 * receive any of those responses to a BYE.
 	 */
-	if (resp >= 400 && resp < 500 && sipmethod == SIP_BYE) {
+	if ((resp == 404 || resp == 408 || resp == 481) && sipmethod == SIP_BYE) {
 		p->needdestroy = 1;
 		return;
 	}