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2011-02-21Merge changes related to AST-2011-002 and FAX-281.lmadsen1-26/+23
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.39.2@308506 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-17AST-2011-001lmadsen1-14/+13
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.39.1@302145 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-07Revert code that changed SSRC for DTMF.jpeeler1-2/+2
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/1.4@297823 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-07Avoid a crash if we don't pass an argument to 'astobj2 test.'seanbright1-0/+4
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@297775 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-02Initialize offset for adaptive jitter buffertwilson1-0/+5
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/1.4@297310 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-01Get rid of the annoying startup and shutdown errors on OS X.tilghman1-0/+12
This mainly deals with the problem of constructors on platforms where an explicit constructor order cannot be specified (any system with gcc 4.2 or less). However, this is only a problem on those systems where we need to initialize mutexes with a constructor, because we have other code that also relies upon constructors, and we cannot specify that mutexes are initialized first (and destroyed last). There are two approaches to dealing with this issue, related to whether the code exists in the core Asterisk binary or in a separate code module. In the core case, constructors are run immediately upon load, and the file_versions list mutex needs to be already initialized, as it is referenced in the first constructor within each core source file. In this case, we use pthread_once to ensure that the mutex is initialized immediately before it is used for the first time. The only caveat is that the mutex is not ever destroyed, but because this is the core, it makes no real difference; the only time when destruction is safe would be just prior to process destruction, which takes care of that anyway. And due to using pthread_once, the mutex will never be reinitialized, which means only one structure has leaked at the end of the process. Hence, it is not a problematic leak. The second approach is to use the load_module and unload_module routines, which, for obvious reasons, exist only in loadable modules. In this second case, we don't have a problem with the constructors, but only with destructor order, because mutexes can be destroyed before their final usage is employed. However, we need the mutexes to still be destroyed, in certain scenarios: if the module is unloaded prior to the process ending, it should be clean, with no allocations by the module hanging around after that point in time. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@296867 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-26Fix bugs in saying numbers using the Swedish language syntaxoej1-38/+45
(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/1.4@296309 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-24Make Asterisk less crashy.russell1-1/+3
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/1.4@296213 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-24Fix false reporting of an error by set_format().russell1-7/+17
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/1.4@296082 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-24Handle failures building translation paths more effectively.russell1-3/+3
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/1.4@296000 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-23Fix support of saynumber(1,n) in the Swedish languageoej1-3/+3
(closes issue #18353) Reported by: oej Review: https://reviewboard.asterisk.org/r/1031/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@295906 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-22The channel redirect function (CLI or AMI) hangs up the call instead of ↵rmudgett2-42/+90
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/1.4@295790 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-16Dead code elimination in channel.c:ast_channel_bridge() variable who.rmudgett1-5/+0
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@295280 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-11I didn't mean to merge this, sorryjpeeler1-3/+0
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@294739 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-11One small addition to 294384 found while very carefully merging to 1.6.jpeeler1-0/+6
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@294641 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-09Fix a deadlock in device state change processing.jpeeler2-107/+192
Copied from some notes from the original author (Russell): Deadlock scenario: Thread 1: device state change thread Holds - rdlock on contexts Holds - hints lock Waiting on channels container lock Thread 2: SIP monitor thread Holds the "iflock" Holds a sip_pvt lock Holds channel container lock Waiting for a channel lock Thread 3: A channel thread (chan_local in this case) Holds 2 channel locks acquired within app_dial Holds a 3rd channel lock it got inside of chan_local Holds a local_pvt lock Waiting on a rdlock of the contexts lock A bunch of other threads waiting on a wrlock of the contexts lock To address this deadlock, some locking order rules must be put in place and enforced. Existing relevant rules: 1) channel lock before a pvt lock 2) contexts lock before hints lock 3) channels container before a channel What's missing is some enforcement of the order when you involve more than any two. To fix this problem, I put in some code that ensures that (at least in the code paths involved in this bug) the locks in (3) come before the locks in (2). To change the operation of thread 1 to comply, I converted the storage of hints to an astobj2 container. This allows processing of hints without holding the hints container lock. So, in the code path that led to thread 1's state, it no longer holds either the contexts or hints lock while it attempts to lock the channels container. (closes issue #18165) Reported by: antonio ABE-2583 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@294384 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-28"!00" evaluated as false, which is incorrect. Fixing.tilghman3-242/+369
Reported (though the reporter did not understand he was reporting a bug) on the asterisk-users list: http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@293194 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-13Don't ignore frames that have been queued when softhangup'dtwilson1-1/+12
When an outgoing call is answered and hung up by the far end *very* quickly, we may not read any frames and therefor end up with a call that displays the wrong disposition/DIALSTATUS. The reason is because ast_queue_hangup() immediately sets the _softhangup flag on the channel and then queues the HANGUP control frame, but __ast_read refuses to read any frames if ast_check_hangup() indicates that a hangup request has been made (which it will if _softhangup is set). So, we end up losing control frames. This change makes __ast_read continue to read frames even if a soft hangup has been requested. It queues a hangup frame to make sure that __ast_read() will still eventually return NULL. Much thanks to David Vossel for all of the reviews, discussion, and help! (closes issue #16946) Reported by: davidw Review: https://reviewboard.asterisk.org/r/740/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@291577 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-12Oops, incorrect range (although unallocated at ARIN)tilghman1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@291263 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-08Ensure editline cleanup occurs when Ctrl-C is pressed at control console.jpeeler1-4/+5
A recent change was made to avoid a race condition on shutdown which only called the end functions from the console thread. However, when pressing Ctrl-C the quit handler is called from the signal handler thread. (closes issue #17698) Reported by: jmls git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@290862 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-02Add documentation for undocumented option to AMI action originateoej1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@289949 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-01Change RFC2833 DTMF event duration on end to report actual elapsed time.jpeeler1-1/+15
The scenario here is with a non P2P early media session. The reported time length of DTMF presses are coming up short when sending to the remote side. Currently the event duration is a running total that is incremented when sending continuation packets. These continuation packets are only triggered upon incoming media from the remote side, which means that the running total probably is not going to end up matching the actual length of time Asterisk received DTMF. This patch changes the end event duration to be lengthened if it is detected that the end event is going to come up short. Review: https://reviewboard.asterisk.org/r/957/ ABE-2476 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@289797 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-29Allow a manager originate to succeed on forwarded devices.qwell1-2/+7
The timeout to wait for an answer was being set to 0 when a device forwarded to another extension. We don't always need the timeout set like this, so make it an optional parameter, and don't use it in this case. ABE-2544 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@289338 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-29Set the caller id on CDRs when it is set on the parent channel.mnicholson1-0/+3
(closes issue #17569) Reported by: tbelder Patches: 17569.diff uploaded by tbelder (license 618) Tested by: tbelder git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@289177 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-28Fixes an issue with the Newchannel AMI event during the Masquerading process.bbryant1-3/+7
Fixes an issue with the Newchannel AMI event during the Masquerading process, where no Newchannel AMI event was generated for the psuedo channel used during the masquerading process. (closes issue #17987) Reported by: RadicAlish Patches: newchannel.patch.txt uploaded by RadicAlish (license 1122) Tested by: RadicAlish Review: https://reviewboard.asterisk.org/r/937/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@289094 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-24Solaris compatibility fixestilghman2-0/+14
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@288636 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-22Fix a 100% CPU consumption problem when setting console=yes in asterisk.conf.russell1-1/+1
The handling of -c and console=yes should be the same, but they were not. When you specify -c, it sets both a flag for console module and for asterisk not to fork() off into the background. The handling of console=yes only set console mode, so you would end up with a background process() trying to run the Asterisk console and freaking out since it didn't have anything to read input from. Thanks to beagles for reporting and helping debug the problem! git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@288339 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-21Add a check to fix a rare segmentation fault you'd get if ast_frdup couldn't ↵bbryant1-1/+3
allocate memory on the first frame being queued in ast_queue_frame. (closes issue #17882) Reported by: seanbright Tested by: seanbright git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@288005 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-21Less than zero is an error, not any non-zero value.tilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@287933 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-20ast_channel_masquerade: remove extra else ifalecdavis1-2/+1
(closes issue #17363,#16057) Reported by: amorsen;davidw,alecdavis Patches: based on bug16057.diff4.txt uploaded by alecdavis (license 585) Tested by: ramonpeek, davidw, alecdavis git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@287684 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-20ast_channel_masquerade: Avoid recursive masquerades.alecdavis1-7/+17
Check all 4 combinations of (original/clonechan) * (masq/masqr). Initially original->masq and clonechan->masqr were only checked. It's possible with multiple masq's planned - and not yet executed, that the 'original' chan could already have another masq'd into it - thus original->masqr would be set, that masqr would lost. Likewise for the clonechan->masq. (closes issue #16057;#17363) Reported by: amorsen;davidw,alecdavis Patches: bug16057.diff4.txt uploaded by alecdavis (license 585) Tested by: ramonpeek, davidw, alecdavis git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@287682 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-20Use ast_dynamic_str when processing hint state changesmnicholson1-6/+10
(related to issue #17928) Reported by: mdu113 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@287555 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-19Make sure we always free variables properly in manager originate.oej1-1/+7
(closes issue #17891) reported, solved and tested by oej Review: https://reviewboard.asterisk.org/r/869/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@287469 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-17Use ast_strdup() instead of ast_strdupa() while processing in ↵mnicholson1-2/+7
ast_hint_state_changed(). (related to issue #17928) Reported by: mdu113 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@287307 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-16Don't limit hint processing in ast_hint_state_changed() to AST_MAX_EXTENSION ↵mnicholson1-3/+1
length strings. (closes issue #17928) Reported by: mdu113 Patches: 20100831__issue17928.diff.txt uploaded by tilghman (license 14) Tested by: mdu113 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@287118 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-16Don't stop printing cdr variables if we encounter one with a blank name or ↵mnicholson1-10/+10
value. (closes issue #17900) Reported by: under Patches: core-show-channel-cdr-fix1.diff uploaded by mnicholson (license 96) Tested by: mnicholson git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@287114 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-14Only drop duplicate answer frames if the channel is bridged.mnicholson1-1/+1
Back in r3710 ast_read() was modified to drop answer frames on channels that were in the UP state. This modification prevented bridges that were up before the answer from being broken and reestablished by an ANSWER control frame. That change also prevents pickup of channels called from the ast_dial framework from working properly. The ast_dial framework expects to see an ANSWER frame after dialing and the pickup code queues one but ast_read() drops it. This new change only drops ANSWER frames when the channel is bridged, allowing the answer queued by the pickup code to properly pass through ast_read() on to the ast_dial framework. ABE-2473 (related to issue #2342) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@286679 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-11Handle error response when we can't make file compatibleoej1-1/+4
Review: https://reviewboard.asterisk.org/r/911/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@286267 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-10Missing newlinetilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@286023 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-09Transmit silence when reading DTMF in ast_readstring.qwell6-193/+278
Otherwise, you could get issues with DTMF timeouts causing hangups. (closes issue #17370) Reported by: makoto Patches: channel-readstring-silence-generator.patch uploaded by makoto (license 38) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@285742 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-29fixes issue with translator frame not getting freeddvossel1-0/+3
A translator frame even if it local storage so the translation path can be freed. This issue prevented g729 licenses from being freed up. (closes issue #17630) Reported by: manvirr Patches: encoder_fix.diff uploaded by dvossel (license 671) Tested by: manvirr, dvossel git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@280448 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-27remove empty audiohook write list on channeldvossel2-1/+21
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/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@279945 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-23Avoid race with consolethread on shutdown (on parallel processors).tilghman1-6/+13
(closes issue #17080) Reported by: sybasesql Patches: 20100721__issue17080.diff.txt uploaded by tilghman (license 14) Tested by: sybasesql git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@278981 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-22Allow PLC to function properly when channels use SLIN for audio.mmichelson1-1/+3
If a channel involved in a bridge was using SLIN audio, then translation paths were not guaranteed to be set up properly since in all likelihood the number of translation steps was only 1. This patch enforces the transcode_via_slin behavior if transcode_via_slin or generic_plc is enabled and one of the formats to make compatible is SLIN. AST-352 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@278618 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-20Do not queue up DTMF frames while a call is on hold.tilghman1-1/+21
(Fixes ABE-2110) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@278167 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-20Off-by-one errortilghman1-5/+3
(closes issue #16506) Reported by: nik600 Patches: 20100629__issue16506.diff.txt uploaded by tilghman (license 14) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@278023 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Interpret device state AST_DEVICE_UNKNOWN as extension state ↵mnicholson1-1/+2
AST_EXTENSION_NOT_INUSE. (closes issue #16035) Reported by: francesco_r Patches: pbx.c.patch uploaded by viniciusfontes (license 978) Tested by: francesco_r, agx, lawbar git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@277327 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16If variable gotten is not set, will segfault on Solaris.tilghman1-1/+1
(closes issue #17636) Reported by: bklang git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@277261 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16For pass through DTMF tones, measure the actual duration between the begin ↵mnicholson1-0/+13
and end packets on the wire. If it is detected to be less than AST_MIN_DTMF_DURATION, trigger dtmf emulation. AST-362 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@277247 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-15In a perfect world, the frame source would never be NULL. In the meantime, ↵jpeeler1-1/+1
don't crash when it is. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@276652 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-01 02:42:20 +0000