Merged revisions 303962 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r303962 | twilson | 2011-01-25 16:09:01 -0600 (Tue, 25 Jan 2011) | 30 lines
  
  Merged revisions 303960 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
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    r303960 | twilson | 2011-01-25 16:02:42 -0600 (Tue, 25 Jan 2011) | 23 lines
    
    Merged revisions 303906 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
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      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/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@303963 f38db490-d61c-443f-a65b-d21fe96a405b

channels/chan_sip.c

@@ -3482,6 +3482,7 @@ static int retrans_pkt(const void *data)
 
 	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);
 		}