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: https://origsvn.digium.com/svn/asterisk/branches/1.8@303962 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index f49c438ce58300668d40896d62b4c5ab86f17649..5fa5bdc669093721648fd2623b90d27557cbbb85 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -3448,6 +3448,7 @@ 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);
                }