Joshua Colp
4a8d87fe98
Add a trigger mode that triggers on both read and write. The actual function that returns the combined audio frame though will wait until both sides have fed in audio, or until one side stops (such as the case when you call Wait).
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(closes issue #11945 )
Reported by: xheliox
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@108083 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-12 18:26:37 +00:00
Russell Bryant
2016f5ba10
Destroy the channel lock after the channel datastores.
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(inspired by issue #12187 )
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@108031 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-12 16:59:07 +00:00
Tilghman Lesher
7c6ceef3ea
Document all of the possible realtime fields
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107877 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-12 01:52:40 +00:00
Jason Parker
c4095ece54
Update documentation for pgsql ODBC voicemail.
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(closes issue #12186 )
Reported by: jsmith
Patches:
vm_pgsql_doc_update.patch uploaded by jsmith (license 15)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107826 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 23:37:05 +00:00
Jason Parker
ea47c2d0b7
Copy voicemail dependency logic for res_adsi to chan_gtalk (for jabber).
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(closes issue #12014 )
Reported by: junky
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107714 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 20:49:56 +00:00
Kevin P. Fleming
d6b2cb9efb
get chan_vpb to build properly in dev mode
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107713 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 20:48:58 +00:00
Jason Parker
48523e3596
Add a newline on a log
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107712 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 20:47:53 +00:00
Joshua Colp
8f65ef8264
Make sure the visible indication is on the right channel so when the masquerade happens the proper indication is enacted.
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(closes issue #11707 )
Reported by: iam
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107646 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 19:20:01 +00:00
Joshua Colp
d9e83f85a6
Add an additional check for setting conference parameter when using the marked user options. It was possible for it to return to a no listen/no talk state if a masquerade happened.
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(closes issue #12136 )
Reported by: aragon
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 18:47:33 +00:00
Joshua Colp
d6d14a3694
Fix a minor spelling error.
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(closes issue #12183 )
Reported by: darrylc
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 17:32:17 +00:00
Kevin P. Fleming
f201a2b11b
backport a fix from trunk
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107472 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 15:18:45 +00:00
Kevin P. Fleming
428a560d33
fix various other problems found by gcc 4.3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107464 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 14:53:03 +00:00
Kevin P. Fleming
9569d74df5
stop checking for mktime() in the configure script... we don't use it, and the test is buggy under gcc 4.3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107461 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 14:33:45 +00:00
Kevin P. Fleming
9f2d2cea4b
check for compiler support for -fno-strict-overflow before using it (tested with Debian's gcc 4.3, 4.1 and 3.4)
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(closes issue #12179 )
Reported by: Netview
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107408 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 14:07:59 +00:00
Kevin P. Fleming
2bfb158faf
fix small bug in IMAP toolkit testing
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107405 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 13:57:08 +00:00
Kevin P. Fleming
553070456b
fix up various compiler warnings found with gcc-4.3:
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- the output of flex includes a static function called 'input' that is not used, so for the moment we'll stop having the compiler tell us about unused variables in the flex source files (a better fix would be to improve our flex post-processing to remove the unused function)
- main/stdtime/localtime.c makes assumptions about signed integer overflow, and gcc-4.3's improved optimizer tries to take advantage of handling potential overflow conditions at compile time; for now, suppress these optimizations until we can fiure out if the code needs improvement
- main/udptl.c has some references to uninitialized variables; in one case there was no bug, but in the other it was certainly possibly for unexpected behavior to occur
- main/editline/readline.c had an unused variable
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107352 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 11:04:29 +00:00
Terry Wilson
28423c15fc
If we fail to alloc a channel, we should re-lock the pvt structure before returning.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107290 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 00:59:18 +00:00
Tilghman Lesher
e1bccfc3fe
Use non-global storage for eswitch
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107230 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-10 21:32:24 +00:00
Jason Parker
be8690e9a8
Make sure to reenable echo can after a "failed" (canceled, etc) three-way call.
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(closes issue #11335 )
Reported by: rebuild
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107173 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-10 20:27:08 +00:00
Russell Bryant
42caaed426
Fix another bug specifically related to asynchronous call origination. Once the
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PBX is started on the channel using ast_pbx_start(), then the ownership of the
channel has been passed on to another thread. We can no longer access it in this
code. If the channel gets hung up very quickly, it is possible that we could
access a channel that has been free'd.
(inspired by BE-386)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107161 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-10 20:17:11 +00:00
Russell Bryant
0a4fc5b8c6
Fix some bugs related to originating calls. If the code failed to start a PBX
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on the channel (such as if you set a call limit based on the system's load
average), then there were cases where a channel that has already been free'd
using ast_hangup() got accessed. This caused weird memory corruption and
crashes to occur.
(fixes issue BE-386)
(much debugging credit goes to twilson, final patch written by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-10 20:04:27 +00:00
Russell Bryant
7a47679898
Resolve a compiler warning.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107102 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-10 17:13:17 +00:00
Russell Bryant
72e4729ad5
Fix a race condition where the generator can go away
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(closes issue #12175 , reported by edantie, patched by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107099 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-10 16:58:57 +00:00
Joshua Colp
2bf8f9fca3
Move where unanswered CDRs are dropped to the CDR core, not everything uses app_dial.
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(closes issue #11516 )
Reported by: ys
Patches:
branch_1.4_cdr.diff uploaded by ys (license 281)
Tested by: anest, jcapp, dartvader
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107016 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-10 14:33:02 +00:00
Kevin P. Fleming
57eaf9dd8f
don't generate D-Channel "up" and "down" messages unless the channel state is actually changing; also, generate the "up" message when an implicit "up" occurs due to reception of a normal event when we thought the channel was "down"
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@106945 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-08 15:59:42 +00:00
Russell Bryant
a04b584272
Only start the SLA thread if SLA has actually been configured.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@106895 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-07 22:51:23 +00:00
Jason Parker
06ba2df183
Fix hardcoded grep in editline, were GNU grep is required.
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(closes issue #12124 )
Reported by: dmartin
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@106842 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-07 22:14:45 +00:00
Joshua Colp
28c66694d3
Ignore source update control frame.
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(closes issue #12168 )
Reported by: plack
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@106788 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-07 19:32:00 +00:00
Russell Bryant
bfed30d5dd
Change a warning message to a debug message. This is happening quite frequently,
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and it is not worth spamming users with these messages unless we are pretty confident
that it should never happen. As it stands today, it _will_ and _does_ happen and
until that gets cleaned up a reasonable amount on the development side, let's not
spam the logs of everyone else.
(closes issue #12154 )
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@106704 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-07 17:16:58 +00:00
Tilghman Lesher
417e8fc754
Warn the user when a temporary greeting exists
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(Closes issue #11409 )
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@106635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-07 16:22:11 +00:00
Tilghman Lesher
072171ef5d
Properly initialize rtp->schedid
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(Closes issue #12154 )
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@106606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-07 15:20:52 +00:00
Tilghman Lesher
56e908b787
Safely use the strncat() function.
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(closes issue #11958 )
Reported by: norman
Patches:
20080209__bug11958.diff.txt uploaded by Corydon76 (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@106552 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-07 06:36:33 +00:00
Mark Michelson
24ca0899c2
Quell an annoying message that is likely to print every single time that
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ast_pbx_outgoing_app is called. The reason is that __ast_request_and_dial
allocates the cdr for the channel, so it should be expected that the channel
will have a cdr on it.
Thanks to joetester on IRC for pointing this out
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@106437 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-06 22:10:07 +00:00
Tilghman Lesher
e0bc9fcf77
Upgrade to the next release of sounds
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@106328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-06 04:40:06 +00:00
Russell Bryant
9479a831f0
Fix a potential deadlock and a few different potential crashes.
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(closes issue #12145 , reported by thiagarcia, patched by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@106237 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-05 22:37:09 +00:00
Joshua Colp
cd703523db
Add a control frame to indicate the source of media has changed. Depending on the underlying technology it may need to change some things.
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(closes issue #12148 )
Reported by: jcomellas
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@106235 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-05 22:32:10 +00:00
Michiel van Baak
ba6b7abe61
document var_metric so no bugreports will come in when it's actually a configuration issue.
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(issue #12151 )
Reported and patched by: caio1982
1.4 patch by me
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@106178 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-05 21:12:36 +00:00
Kevin P. Fleming
461e3fea79
when a PRI call must be moved to a different B channel at the request of the other endpoint, ensure that any DSP active on the original channel is moved to the new one
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(closes issue #11917 )
Reported by: mavetju
Tested by: mavetju
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@106038 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-05 15:32:35 +00:00
Tilghman Lesher
b350a97937
Correctly initialize retransid in SIP, and ensure that the warning when failing to delete a schedule entry can actually hit the log.
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(closes issue #12140 )
Reported by: slavon
Patches:
sch2.patch uploaded by slavon (license 288)
(Patch slightly modified by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@106015 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-05 15:17:16 +00:00
Russell Bryant
d564404d73
Fix a bug that I just noticed in the RTP code. The calculation for setting the
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len field in an ast_frame of audio was wrong when G.722 is in use. The len field
represents the number of ms of audio that the frame contains. It would have
set the value to be twice what it should be.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@105932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-05 01:52:18 +00:00
Joshua Colp
be005c60d6
In addition to setting the marker bit let's change our ssrc so they know for sure it is a different source.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@105676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-04 18:10:34 +00:00
Joshua Colp
36bb1f9d46
When a new source of audio comes in (such as music on hold) make sure the marker bit gets set.
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(closes issue #10355 )
Reported by: wdecarne
Patches:
10355.diff uploaded by file (license 11)
(closes issue #11491 )
Reported by: kanderson
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@105674 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-04 18:05:28 +00:00
Russell Bryant
40425a24bf
Backport a minor bug fix from trunk that I found while doing random code
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cleanup. Properly break out of the loop when a context isn't found when
verify that includes are valid.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@105591 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-04 04:31:29 +00:00
Jason Parker
47f33c87f4
Fix type for astNumChannels.
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(closes issue #12114 )
Reported by: jeffg
Patches:
12114.patch uploaded by jeffg (license 192)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@105572 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-03 18:06:52 +00:00
Russell Bryant
7f7dbcb11f
In the case of an ast_channel allocation failure, take the local_pvt out of the
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pvt list before destroying it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@105570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-03 17:16:53 +00:00
Russell Bryant
b3c0e042d4
Fix a potential memory leak of the local_pvt struct when ast_channel allocation
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fails. Also, in passing, centralize the code necessary to destroy a local_pvt.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@105568 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-03 17:05:16 +00:00
Russell Bryant
9c5af082ca
Update the copyright information for autoservice. Most of the code in this file
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now is stuff that I have written recently ...
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@105565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-03 16:01:50 +00:00
Russell Bryant
547ac9f501
Merge in some changes from team/russell/autoservice-nochans-1.4
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These changes fix up some dubious code that I came across while auditing what
happens in the autoservice thread when there are no channels currently in
autoservice.
1) Change it so that autoservice thread doesn't keep looping around calling
ast_waitfor_n() on 0 channels twice a second. Instead, use a thread condition
so that the thread properly goes to sleep and does not wake up until a
channel is put into autoservice.
This actually fixes an interesting bug, as well. If the autoservice thread
is already running (almost always is the case), then when the thread goes
from having 0 channels to have 1 channel to autoservice, that channel would
have to wait for up to 1/2 of a second to have the first frame read from it.
2) Fix up the code in ast_waitfor_nandfds() for when it gets called with no
channels and no fds to poll() on, such as was the case with the previous code
for the autoservice thread. In this case, the code would call alloca(0), and
pass the result as the first argument to poll(). In this case, the 2nd
argument to poll() specified that there were no fds, so this invalid pointer
shouldn't actually get dereferenced, but, this code makes it explicit and
ensures the pointers are NULL unless we have valid data to put there.
(related to issue #12116 )
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@105563 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-03 15:50:43 +00:00
Joshua Colp
5bfdc9818d
It is possible for no audio to pass between the current digit and next digit so expand logic that clears emulation to AST_FRAME_NULL.
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(closes issue #11911 )
Reported by: edgreenberg
Patches:
v1-11911.patch uploaded by dimas (license 88)
Tested by: tbsky
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@105560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-03 15:28:59 +00:00
Joshua Colp
70d43ff1d2
Add a comment to describe some logic.
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(closes issue #12120 )
Reported by: flefoll
Patches:
chan_sip.c.br14.patch-just-a-comment uploaded by flefoll (license 244)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@105557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-03 15:15:39 +00:00