Age | Commit message (Collapse) | Author | Files | Lines |
|
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.16.2@308524 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.16.2@308515 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.16.2@308510 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.16.2@308507 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.16.2@308504 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.16.1@302156 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.16.1@302148 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.16.1@302093 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.16@301498 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.16@301493 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
------------------------------------------------------------------------
r301220 | pabelanger | 2011-01-09 15:38:25 -0600 (Sun, 09 Jan 2011) | 14 lines
SOUND_CACHE_DIR now defaults to empty
Sounds files included in the Asterisk tarball were being ignored and
re-downloaded. Users wanting to cache the files can still override the setting
using the --with-sounds-cache option.
(closes issue 0018589)
Reported by: pabelanger
Patches:
issue18589.patch uploaded by pabelanger (license 224)
Tested by: pabelanger
Review: https://reviewboard.asterisk.org/r/1074/ [^]
------------------------------------------------------------------------
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.16@301492 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.16@301460 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.16-rc1@298187 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.16-rc1@298186 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.16-rc1@298185 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.16-rc1@298184 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Also detect the required structure element, because OpenSolaris defines
SIOCGIFHWADDR, but without support for IP sockets.
(closes issue #18442)
Reported by: ranjtech
Patches:
20101209__issue18442.diff.txt uploaded by tilghman (license 14)
Tested by: ranjtech
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@298050 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r297959 | twilson | 2010-12-09 16:00:30 -0600 (Thu, 09 Dec 2010) | 14 lines
Ignore spurious REGISTER requests
If a REGISTER request with a Call-ID matching an existing transaction is received
it was possible that the REGISTER request would overwrite the initreq of the
private structure. This info is used to generate messages for other responses in
the transaction. This patch ignores REGISTER requests that match non-REGISTER
transactions.
(closes issue #18051)
Reported by: eeman
Tested by: twilson
Review: https://reviewboard.asterisk.org/r/1050/
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@297960 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(from an internal Digium discussion)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@297908 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r297823 | jpeeler | 2010-12-07 16:57:48 -0600 (Tue, 07 Dec 2010) | 12 lines
Revert code that changed SSRC for DTMF.
Some previous behavior was attempted to be restored, but mistakingly I did
not realize that the previous behavior was incorrect. This fixes DTMF not
being detected since DTMF shouldn't cause the SSRC to change.
(related to issue #17404)
(closes issue #18189)
(closes issue #18352)
Reported by: marcbou
Tested by: cmbaker82
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@297824 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r297818 | tilghman | 2010-12-07 16:35:50 -0600 (Tue, 07 Dec 2010) | 4 lines
Use non-deprecated APIs for CoreAudio
Review: https://reviewboard.asterisk.org/r/1040/
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@297819 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r297689 | tilghman | 2010-12-06 18:07:37 -0600 (Mon, 06 Dec 2010) | 8 lines
Don't create a Local channel if the target extension does not exist.
(closes issue #18126)
Reported by: junky
Patches:
followme.diff uploaded by junky (license 177)
(partially restructured by me to avoid a possible memory leak)
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@297713 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r297603 | jpeeler | 2010-12-06 15:57:15 -0600 (Mon, 06 Dec 2010) | 12 lines
Improve handling of REGISTER requests with multiple contact headers.
The changes here attempt to more strictly follow RFC 3261 section 10.3.
Basically the following will now cause a 400 Bad Response to be returned, if:
- multiple Contact headers are present with one set to expire all bindings ("*")
- wildcard parameter is specified for Contact without Expires header or Expires
header is not set to zero.
ABE-2442
ABE-2443
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@297605 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@297534 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r297404 | pabelanger | 2010-12-02 15:01:08 -0500 (Thu, 02 Dec 2010) | 7 lines
Resolve compile error under FreeBSD
We now set _ASTCFLAGS+=-march=i686 for i386 processors, still allowing ASTCFLAGS
to override the setting.
Review: https://reviewboard.asterisk.org/r/1043/
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@297405 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r297310 | twilson | 2010-12-02 12:00:27 -0600 (Thu, 02 Dec 2010) | 12 lines
Initialize offset for adaptive jitter buffer
When the adaptive jitter buffer is enabled in sip.conf, the first frame placed
in the jitter buffer fails with something like:
jb_warning_output: Resyncing the jb. last_delay 0, this delay -215886466,
threshold 1000, new offset 215886466
This happens because the offset is not initialized before calling jb_put(). This
patch modifies jb_put_first_adaptive() to set the offset to the frame's
timestamp.
Review: https://reviewboard.asterisk.org/r/1041/
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@297311 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r297228 | russell | 2010-12-02 07:16:15 -0600 (Thu, 02 Dec 2010) | 6 lines
Add "DAHDI" to a couple of app_meetme error messages.
This is in response to some questions on IRC. To the user, there was nothing
that made it obvious that this error had anything to do with DAHDI not being
loaded.
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@297229 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r297185 | oej | 2010-12-02 09:37:17 +0100 (Tor, 02 Dec 2010) | 5 lines
If we get a NOTIFY from a non-existing subscription we should answer with 481, not bad event.
If we answer 481 the subscription that we don't want will be cancelled.
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@297186 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r297072 | jpeeler | 2010-12-01 11:50:09 -0600 (Wed, 01 Dec 2010) | 23 lines
Fix not stopping MOH when transfered local channel queue member is answered.
The problem here is only present when local channels are used with the MOH
passthru option as well as no optimization (/nm). I will describe the slightly
bizarre scenario that was used to test, where phones B and C are queue members:
Phone A dials into a queue with two members using local channels and the above
options. Phone B answers. Phone A blind transfers phone B into the same queue.
Phone A hangs up. Phone C answers, but phone B didn't stop playing MOH.
In this scenario, the unhold frame that should have gotten to phone B never
arrived due to the masquerade from the blind transfer. This is usually fine
since app_queue manages the starting and stopping of MOH. However, with the
passthrough option enabled when app_queue attempts to stop MOH it tries to do
so on the local channel rather than the real channel. The easiest solution
was to just make sure to send an unhold frame during the transfer since it
wouldn't make sense to have MOH playing after a transfer anyway. This only
modifies SIP transfers, but the other transfers did not seem to be a problem.
If DTMF based transfers were a problem it might be okay to add ast_moh_stop
to finishup, but I didn't want to have to add that unless required.
ABE-2624
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@297073 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r296990 | tilghman | 2010-12-01 10:59:26 -0600 (Wed, 01 Dec 2010) | 5 lines
Clarify documentation on how we store codec preference lists.
(closes issue #18397)
Reported by: birgita
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@296991 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
others, too).
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@296950 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r296868 | jpeeler | 2010-11-30 18:23:19 -0600 (Tue, 30 Nov 2010) | 4 lines
Properly restore backup information file when hanging up during message prepending.
ABE-2654
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@296869 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r296670 | pabelanger | 2010-11-29 17:49:39 -0500 (Mon, 29 Nov 2010) | 5 lines
Make sure nothing else is needed before destroying the scheduler.
(closes issue #18398)
Reported by: pabelanger
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@296671 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
isn't one.
Linux and *BSD disagree on the elements within the ucred structure. Detect
which one is in use on the system.
(closes issue #18384)
Reported by: bjm
Patches:
cred-diffs uploaded by bjm (license 473)
20101127__issue18384__1.6.2.diff.txt uploaded by tilghman (license 14)
20101127__issue18384__1.8.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman, bjm
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@296533 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
more in case of greater precision).
(closes issue #18369)
Reported by: tnakonz
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@296466 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r296309 | oej | 2010-11-26 10:53:31 +0100 (Fre, 26 Nov 2010) | 11 lines
Fix bugs in saying numbers using the Swedish language syntax
(closes issue #18355)
Reported by: oej
Patch by: oej
Much help from Peter Lindahl. Testing by the ClearIT team during a coffee break.
Review: https://reviewboard.asterisk.org/r/1033/
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@296351 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r296213 | russell | 2010-11-24 17:26:43 -0600 (Wed, 24 Nov 2010) | 6 lines
Make Asterisk less crashy.
Since we might not put a new translation path on the channel, go ahead and
set it to NULL right after destroying the old one to ensure we don't try
to free an invalid translation path later on.
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@296221 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r296165 | rmudgett | 2010-11-24 16:41:07 -0600 (Wed, 24 Nov 2010) | 43 lines
Oneway audio to SIP phone from FXS port after FXS port gets a CallWaiting pip.
The FXS connected phone has to have CW/CID support to fail, as it will
send back a DTMF 'A' or 'D' when it's ready to receive CallerID. A normal
phone with no CID never fails. Also the SIP phone does not hear MOH when
the CW call is answered.
The DTMF end frame is suppressed when the phone acknowledges the CW signal
for CID. The problem is the DTMF begin frame needs to be suppressed as
well. The DTMF begin frame is causing SIP to start sending the DTMF RTP
frames. Since the DTMF end frame is suppressed, SIP will not stop sending
those DTMF RTP packets.
* Suppress the DTMF begin and end frames when the channel driver is
looking for DTMF digits.
* Fixed a couple issues caused by not cleaning up the CID spill if you
answer the CW call while it is sending the CID spill.
* Fixed not sending CW/CID spill to the phone when the call is natively
bridged. (Fixed by not using native bridge if CW/CID is possible.)
* Suppress received audio when sending CW/CID spills. The other parties
involved do not need to hear the CW/CID spills and may be confused if the
CW call is for them.
(closes issue #18129)
Reported by: alecdavis
Patches:
issue_18129_v1.8_v3.patch uploaded by rmudgett (license 664)
Tested by: alecdavis, rmudgett
NOTE:
* v1.4 does not have the main problem fixed by suppressing the DTMF start
frames. The other three items fixed are relevant.
* If you really must restore native bridging between analog ports, you
need to disable CW/CID either by configuring chan_dahdi.conf
callwaitingcallerid=no or dialing *70 before dialing the number to
temporarily disable CW.
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@296166 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r296082 | russell | 2010-11-24 14:22:32 -0600 (Wed, 24 Nov 2010) | 12 lines
Fix false reporting of an error by set_format().
In the case that the native format was able to be changed to match the
new requested format, the code proceeded to attempt to build a translation
path, anyway. The result would be NULL, since no translation path is
necessary and resulted in this function thinking an error has occurred.
This case is now specifically caught and no attempt to build a translation
path is attempted.
Thanks to our automated tests and bamboo.asterisk.org for catching this problem
and making a whole lot of noise when things started failing. :-)
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@296083 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010) | 38 lines
Handle failures building translation paths more effectively.
The problem scenario occurred on a heavily loaded system that was using the
codec_dahdi module and exceeded the hardware transcoding capacity. The failure
mode at that point was not good. The report came in to us as an Asterisk
lock-up. The "core show locks" shows a ton of threads locked up (but no
obvious deadlock). Upon deeper investigation, when the system is in this
state, the CPU was maxed out. The CPU was being consumed by the Asterisk
logger spewing messages on every audio frame for calls set up after transcoder
capacity was reached.
The purpose of this patch is to make Asterisk handle failures to create a
translation path in a more graceful manner. If we can't translate, then the
call just needs to be dropped, as it's not going to work. These are the
changes:
1) In set_format() of channel.c (which is called by set_read_format() and
set_write_format()), it was ignoring if ast_translator_build_path() failed and
returned NULL. It now pays attention to that case and returns a result
reflecting failure. With this change in place, the bridging code will
immediately detect a failure and end the bridge instead of proceeding to try to
bridge frames that can't be translated and making channel drivers freak out by
sending them frames in a format they weren't expecting.
2) In ast_indicate_data() of channel.c, failure of ast_playtones_start() was
ignored. It is now reflected in the return value of the function. This didn't
turn out to have any affect on the bug, but seemed like a good change to leave
in.
3) In app_dial(), when only sending a call to a single endpoint, it will
attempt to do some bridging of its own of early audio. It uses
make_compatible() when it's going to do this. However, it ignored failure from
make compatible. So, even with the fix from #1, if there was early audio going
through app_dial, there would still be a period of invalid frames passing
through. After detecting failure here, Dial() exits.
ABE-2658
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@296001 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r295906 | oej | 2010-11-23 10:28:14 +0100 (Tis, 23 Nov 2010) | 8 lines
Fix support of saynumber(1,n) in the Swedish language
(closes issue #18353)
Reported by: oej
Review: https://reviewboard.asterisk.org/r/1031/
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@295907 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@295868 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010) | 46 lines
The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call.
To recreate the problem:
1) Party A calls Party B
2) Invoke CLI "channel redirect" command to redirect channel call leg
associated with A.
3) All associated channels are hung up.
Note that if the CLI command were done on the channel call leg associated
with B it works.
This regression was a result of the fix for issue #16946
(https://reviewboard.asterisk.org/r/740/).
The regression affects all features that use an async goto to execute the
dialplan because of an external event: Channel redirect, AMI redirect, SIP
REFER, and FAX detection.
The struct ast_channel._softhangup code is a mess. The variable is used
for several purposes that do not necessarily result in the call being hung
up. I have added doxygen comments to describe how the various _softhangup
bits are used. I have corrected all the places where the variable was
tested in a non-bit oriented manner.
The primary fix is the new AST_CONTROL_END_OF_Q frame. It acts as a weak
hangup request so the soft hangup requests that do not normally result in
a hangup do not hangup.
JIRA SWP-2470
JIRA SWP-2489
(closes issue #18171)
Reported by: SantaFox
(closes issue #18185)
Reported by: kwemheuer
(closes issue #18211)
Reported by: zahir_koradia
(closes issue #18230)
Reported by: vmarrone
(closes issue #18299)
Reported by: mbrevda
(closes issue #18322)
Reported by: nerbos
Review: https://reviewboard.asterisk.org/r/1013/
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@295843 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
This patch addresses a regression where device states across multiple servers
were not being processing completely correctly. The code works to determine
the overall state by looking at the last known state of a device on each
server. However, there was a regression due to some invasive rewrites of how
the cache works that led to the cache only storing the last device state change
for a device, regardless of which server it was on.
The code is set up to cache device state change events by ensuring that each
event in the cache has a unique device name + entity ID (server ID). The code
that was responsible for comparing raw information elements (which EID is)
always returned a match due to a memcmp() with a length of 0.
There isn't much code to fix the actual bug. This patch also introduces a new
CLI command that was very useful for debugging this problem. The command
allows you to dump the contents of the event cache.
(closes issue #18284)
Reported by: klaus3000
Patches:
issue18284.rev1.txt uploaded by russell (license 2)
Tested by: russell, klaus3000
(closes issue #18280)
Reported by: klaus3000
Review: https://reviewboard.asterisk.org/r/1012/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@295710 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r295628 | twilson | 2010-11-19 12:53:36 -0800 (Fri, 19 Nov 2010) | 8 lines
Discard responses with more than one Via
This is not a perfect solution as headers that are joined via commas are not
detected. This is a parsing issue that to fix "correctly" would necessitate
a new SIP parser.
Review: https://reviewboard.asterisk.org/r/1019/
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@295672 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Review: https://reviewboard.asterisk.org/r/1016/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@295440 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r295280 | rmudgett | 2010-11-16 16:52:06 -0600 (Tue, 16 Nov 2010) | 1 line
Dead code elimination in channel.c:ast_channel_bridge() variable who.
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@295281 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r295026 | tilghman | 2010-11-15 11:58:37 -0600 (Mon, 15 Nov 2010) | 2 lines
Create test verifying results of expression parser
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@295062 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(closes issue #18161)
Reported by: wdoekes
Patches:
20101029__issue18161.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@294988 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
open_mailbox actually caused it to be fixed, but let's be consistent.
Reported by alecdavis in asterisk-dev.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@294910 f38db490-d61c-443f-a65b-d21fe96a405b
|