[svn-commits] twilson: branch 1.6.2 r303960 - in /branches/1.6.2: ./ channels/chan_sip.c

SVN commits to the Digium repositories svn-commits at lists.digium.com
Tue Jan 25 16:02:47 CST 2011


Author: twilson
Date: Tue Jan 25 16:02:42 2011
New Revision: 303960

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=303960
Log:
Merged revisions 303906 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r303906 | twilson | 2011-01-25 14:50:59 -0600 (Tue, 25 Jan 2011) | 16 lines
  
  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/1.6.2/   (props changed)
    branches/1.6.2/channels/chan_sip.c

Propchange: branches/1.6.2/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.

Modified: branches/1.6.2/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.2/channels/chan_sip.c?view=diff&rev=303960&r1=303959&r2=303960
==============================================================================
--- branches/1.6.2/channels/chan_sip.c (original)
+++ branches/1.6.2/channels/chan_sip.c Tue Jan 25 16:02:42 2011
@@ -3869,6 +3869,7 @@
 
 	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.");




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