From 3725173b9e18374e84af2fed59c245d5d15eb4bb Mon Sep 17 00:00:00 2001 From: Matthew Jordan Date: Thu, 26 Feb 2015 03:03:39 +0000 Subject: 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 c743e0f42..cf2cd097c 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); -- cgit v1.2.3