[svn-commits] twilson: branch 1.4 r303906 - /branches/1.4/channels/chan_sip.c

[SVN commits to the Digium repositories]

Author: twilson
Date: Tue Jan 25 14:50:59 2011
New Revision: 303906

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=303906
Log:
Guard against retransmitting BYEs indefinitely

In the case of an attended transfer (A calls B, A atxfers to C) where
A becomes unreachable before replying to Asterisk's BYE, Asterisk can
sometimes retransmit the BYE indefinitely. This is because
__sip_autodestruct tests p->refer && !ast_test_flag(&p->flags[0],
SIP_ALREADYGONE and will then transmit a BYE. When this BYE times out,
it will not ever be marked as ALREADYGONE, so when __sip_autodestruct
is called again, we end up starting the cycle over.

This patch adds a call to sip_alreadygone(pkt->owner) in retrans_pkt
in the case of a BYE that has timed out. This should prevent Asterisk
from trying to transmit new BYE messages in the future.

Review: https://reviewboard.asterisk.org/r/1077/

Modified:
    branches/1.4/channels/chan_sip.c

Modified: branches/1.4/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=303906&r1=303905&r2=303906
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Tue Jan 25 14:50:59 2011
@@ -2094,6 +2094,7 @@
 
 	if (pkt->method == SIP_BYE) {
 		/* We're not getting answers on SIP BYE's.  Tear down the call anyway. */
+		sip_alreadygone(pkt->owner);
 		if (pkt->owner->owner)
 			ast_channel_unlock(pkt->owner->owner);
 		append_history(pkt->owner, "ByeFailure", "Remote peer doesn't respond to bye. Destroying call anyway.");