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authortwilson <twilson@f38db490-d61c-443f-a65b-d21fe96a405b>2011-01-25 20:50:59 +0000
committertwilson <twilson@f38db490-d61c-443f-a65b-d21fe96a405b>2011-01-25 20:50:59 +0000
commitd4c987b707a372cf8a31ea5fb7d9eea1bae19cb5 (patch)
treeec91edaed2ab9b3ffd713b8849b749ed837e02f6
parent76931c67d3aeead7e37cabdfb04ca93debf9099a (diff)
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: http://svn.digium.com/svn/asterisk/branches/1.4@303906 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r--channels/chan_sip.c1
1 files changed, 1 insertions, 0 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 374af6174..55cb54a82 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.");