[asterisk-commits] kmoore: trunk r331519 - in /trunk: ./ channels/chan_sip.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Aug 10 17:24:42 CDT 2011
Author: kmoore
Date: Wed Aug 10 17:24:38 2011
New Revision: 331519
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=331519
Log:
Merged revisions 331518 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................
r331518 | kmoore | 2011-08-10 17:23:49 -0500 (Wed, 10 Aug 2011) | 17 lines
Merged revisions 331517 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r331517 | kmoore | 2011-08-10 17:23:08 -0500 (Wed, 10 Aug 2011) | 10 lines
SIP Notify via AMI or CLI leaks SIP PVTs
Any SIP notify sent via AMI or CLI leaks a SIP PVT with ref count +2. Removing
the additional ref just before the invite and adding an unref following it
corrects the issue as seen via REF_DEBUG. The unref existed in a distant
revision and it appears as though the wrong ref operation was removed.
(closes issue ASTERISK-18091)
Review: https://reviewboard.asterisk.org/r/1332/
........
................
Modified:
trunk/ (props changed)
trunk/channels/chan_sip.c
Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-10-merged' - no diff available.
Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=331519&r1=331518&r2=331519
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Wed Aug 10 17:24:38 2011
@@ -12744,9 +12744,9 @@
}
}
- dialog_ref(p, "bump the count of p, which transmit_sip_request will decrement.");
sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
transmit_invite(p, SIP_NOTIFY, 0, 2, NULL);
+ dialog_unref(p, "bump down the count of p since we're done with it.");
astman_send_ack(s, m, "Notify Sent");
ast_variables_destroy(vars);
@@ -18966,9 +18966,9 @@
/* Recalculate our side, and recalculate Call ID */
ast_cli(a->fd, "Sending NOTIFY of type '%s' to '%s'\n", a->argv[2], a->argv[i]);
- dialog_ref(p, "bump the count of p, which transmit_sip_request will decrement.");
sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
transmit_invite(p, SIP_NOTIFY, 0, 2, NULL);
+ dialog_unref(p, "bump down the count of p since we're done with it.");
}
return CLI_SUCCESS;
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