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Jon Bonilla (Manwe) pointed out on the -dev list:

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"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: https://origsvn.digium.com/svn/asterisk/branches/1.4@160480 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Tilghman Lesher
2008-12-03 14:09:35 +00:00
parent de2bff02a1
commit cc3b3e68f0
1 changed files with 5 additions and 6 deletions
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channels/chan_sip.c
+5 -6
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@@ -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;
}