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.


git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@160480 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 5 insertions, 6 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index a98afc3d9..26fcec09a 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -12873,13 +12873,12 @@ 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) {
 		ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
 		return;
 	}