diff options
| author | twilson <twilson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2011-01-25 22:09:01 +0000 |
|---|---|---|
| committer | twilson <twilson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2011-01-25 22:09:01 +0000 |
| commit | 75b6fe8aee0d3a065ed1e87aa3cad3118177a296 (patch) | |
| tree | baa2f5b5096c1b99bf42a8f233cf3a05ebe7a044 /channels/chan_sip.c | |
| parent | d8bd03cc61d7365cf09c9b0bec0b12a1a4567924 (diff) | |
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/branches/1.8@303962 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels/chan_sip.c')
| -rw-r--r-- | channels/chan_sip.c | 1 |
1 files changed, 1 insertions, 0 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index f49c438ce..5fa5bdc66 100644 --- 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); } |