https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r281430 | dvossel | 2010-08-09 15:46:50 -0500 (Mon, 09 Aug 2010) | 13 lines
fixes SIP peers memory leak
We zeroed out the peer's addr before it was removed from the
peers_by_ip container. This made it impossible to be removed
from the container as the addr is the key used by the container
to find the peer.
(closes issue #17774)
Reported by: kkm
Patches:
017774-sip-peer-leak-1.6.2.10.diff uploaded by kkm (license 888)
017774-sip-peer-leak-1.8.diff uploaded by kkm (license 888)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@281432 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r281391 | jpeeler | 2010-08-09 15:07:29 -0500 (Mon, 09 Aug 2010) | 20 lines
Merged revisions 281390 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r281390 | jpeeler | 2010-08-09 15:04:30 -0500 (Mon, 09 Aug 2010) | 13 lines
Prevent loss of Caller ID information set on local channel after masquerade.
Caller ID set on the channel before a masquerade occurs when using a local
channel would cause the information to be lost. The problem was that the
information was set on a channel destined to be hung up. The somewhat confusing
fix is to detect if any Caller ID has been set on the channel and if so
preswap the Caller ID data so that basically the masquerade puts the data back.
(closes issue #17138)
Reported by: kobaz
Review: https://reviewboard.asterisk.org/r/847/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@281429 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Put all of the options that affect the contents of CDRs together, instead
of having the batch mode options in the middle of them.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@281294 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r281051 | russell | 2010-08-05 08:11:32 -0500 (Thu, 05 Aug 2010) | 9 lines
Cleanup default option value handling for cdr.conf [general].
The default values would differ depending on whether or not cdr.conf exists.
That is no longer the case.
Apply a default value to the unanswered option.
Define all default values as named constants.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@281052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r280227 | qwell | 2010-07-28 14:54:54 -0500 (Wed, 28 Jul 2010) | 1 line
Add sha1sum-sh in case there is no util on the system.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@280232 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The version of libedit that is bundled with asterisk is old and has some bugs.
This patch updates the bundled version of libedit within asterisk, and also
updates asterisk to use the system libedit instead if one is available (and
pkg-config is available). This review integrates several patches from other
users specifically kkm and tzafrir.
(closes issue #15929)
Reported by: kkm
Patches:
015929-astcli-editrc-trunk.240324.diff uploaded by kkm (license 888)
(issue #16858)
Reported by: jw-asterisk
(closes issue #17039)
Reported by: tzafrir
Patches:
0001-allow-using-system-copy-of-libedit.patch uploaded by tzafrir (license 46)
Review: https://reviewboard.asterisk.org/r/807/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@280019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r279946 | dvossel | 2010-07-27 15:54:32 -0500 (Tue, 27 Jul 2010) | 24 lines
Merged revisions 279945 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010) | 19 lines
remove empty audiohook write list on channel
If a channel has an audiohook write list created on it, that
list stays on the channel until the channel is destroyed. There
is no reason to keep that list on the channel if it becomes empty.
If it is empty that just means we are doing needless translating
for every ast_read and ast_write. This patch removes the audiohook
list from the channel once it is detected to be empty on either a
read or write. If a audiohook is added back to the channel after
this list is destroyed, the list just gets recreated as if it never
existed to begin with.
(closes issue #17630)
Reported by: manvirr
Review: https://reviewboard.asterisk.org/r/799/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279949 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is a regression from the sig_pri split from chan_dahdi. When a call is
first initiated, the inband DTMF detector is not enabled if it's an outgoing
ISDN call. However, it needs to be turned on once the media path starts up.
This handling was put back in the open_media() callback of chan_dahdi. In
sig_pri, open_media() calls were added to a few places where it was needed,
including handling of PRI_EVENT_RINGING, PRI_EVENT_PROGRESS, and
PRI_EVENT_PROCEEDING.
Thanks to rmudgett for helping me with the patch!
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279916 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The code was written in a way that did a bad job of
parsing the port out of a URI. Specifically, it would
do badly when dealing with an IPv6 address. In this
particular scenario, there was no value from parsing
the port out, so I just removed that logic. And while
I was messing around in the function, I changed some
variable names to be more descriptive.
(closes issue #17661)
Reported by: oej
Patches:
17661.diff uploaded by mmichelson (license 60)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r279883 | qwell | 2010-07-27 12:54:54 -0500 (Tue, 27 Jul 2010) | 1 line
Add SHA1SUM to configure, since we require it for sounds/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279884 65c4cc65-6c06-0410-ace0-fbb531ad65f3
