From 9f262f2772045266b309c00598c498139ad332b0 Mon Sep 17 00:00:00 2001 From: Terry Wilson Date: Tue, 25 Jan 2011 20:50:59 +0000 Subject: [PATCH] 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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@303906 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- channels/chan_sip.c | 1 + 1 file changed, 1 insertion(+) diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 374af61740..55cb54a820 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -2094,6 +2094,7 @@ 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); append_history(pkt->owner, "ByeFailure", "Remote peer doesn't respond to bye. Destroying call anyway.");