From 3725173b9e18374e84af2fed59c245d5d15eb4bb Mon Sep 17 00:00:00 2001
From: Matthew Jordan <mjordan@digium.com>
Date: Thu, 26 Feb 2015 03:03:39 +0000
Subject: [PATCH] channels/chan_sip: Don't send a BYE after final response when
PBX thread fails
When Asterisk fails to start a PBX thread for a new channel - for example, when
the maxcalls setting in asterisk.conf is exceeded - we currently send a final
response, and then attempt to send a BYE request to the UA. Since that's all
sorts of wrong, this patch fixes that by setting sipalreadygone on the sip_pvt
such that we don't get stuck sending BYE requests to something that does not
want it.
Note that this patch is a slight modification of the one on ASTERISK-15434.
For clarity, it explicitly calls sipalreadygone with the calls to transmit a
final response.
ASTERISK-21845
ASTERISK-15434 #close
Reported by: Makoto Dei
Tested by: Matt Jordan
patches:
sip-pbxstart-failed.patch uploaded by Makoto Dei (License 5027)
........
Merged revisions 432320 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 432321 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432322 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
channels/chan_sip.c | 2 ++
1 file changed, 2 insertions(+)
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index c743e0f426..cf2cd097c3 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -25819,11 +25819,13 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, str
switch(result) {
case AST_PBX_FAILED:
+ sip_alreadygone(p);
ast_log(LOG_WARNING, "Failed to start PBX :(\n");
p->invitestate = INV_COMPLETED;
transmit_response_reliable(p, "503 Unavailable", req);
break;
case AST_PBX_CALL_LIMIT:
+ sip_alreadygone(p);
ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
p->invitestate = INV_COMPLETED;
transmit_response_reliable(p, "480 Temporarily Unavailable", req);
--
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