[svn-commits] mjordan: branch 13 r432321 - in /branches/13: ./ channels/chan_sip.c

[SVN commits to the Digium repositories]

Author: mjordan
Date: Wed Feb 25 21:03:06 2015
New Revision: 432321

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=432321
Log:
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

Modified:
    branches/13/   (props changed)
    branches/13/channels/chan_sip.c

Propchange: branches/13/
------------------------------------------------------------------------------
Binary property 'branch-11-merged' - no diff available.

Modified: branches/13/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/13/channels/chan_sip.c?view=diff&rev=432321&r1=432320&r2=432321
==============================================================================
--- branches/13/channels/chan_sip.c (original)
+++ branches/13/channels/chan_sip.c Wed Feb 25 21:03:06 2015
@@ -25806,11 +25806,13 @@
 
 				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);