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r294384 | jpeeler | 2010-11-09 11:37:59 -0600 (Tue, 09 Nov 2010) | 47 lines
Fix a deadlock in device state change processing.
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
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(closes issue #16757)
Reported by: voxter
Patches:
20101012__issue16757.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/994/
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(closes issue #18282)
Reported by: klaus3000
Patches:
ast_devstate2str-patch.txt uploaded by klaus3000 (license 65)
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To fix this issue the alert pipe will now be used when the timerfd module is
in use. There appeared to be a race that was not solved by adding locking in the
timerfd module, but needed to be there anyway. The race was between the timer
being put in non-continuous mode in ast_read on the channel thread and the IAX
frame scheduler queuing a frame which would enable continuous mode before the
non-continuous mode event was read. This race for now is simply avoided.
(closes issue #18110)
Reported by: tpanton
Tested by: tpanton
I put tested by tpanton because it was tested on his hardware. Thanks for the
remote access to debug this issue!
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r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010) | 5 lines
"!00" evaluated as false, which is incorrect. Fixing.
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
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r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010) | 5 lines
"!00" evaluated as false, which is incorrect. Fixing.
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
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r291577 | twilson | 2010-10-13 15:45:15 -0700 (Wed, 13 Oct 2010) | 21 lines
Don't ignore frames that have been queued when softhangup'd
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/
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r291263 | tilghman | 2010-10-12 11:55:30 -0500 (Tue, 12 Oct 2010) | 2 lines
Oops, incorrect range (although unallocated at ARIN)
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Setting the module/filename specific message level and then changing it
resulted in the linked list being looped on itself. Traversing this
linked list is an infinite loop if what you are looking for is not in the
list.
Also plugged some CLI parsing holes in the associated CLI command:
* Removing a nonexistent module from the list actually added it with a
level of zero.
* Setting the non-module specific level to zero is now equivalent to
setting it to "off" as documented.
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r290862 | jpeeler | 2010-10-07 21:35:29 -0500 (Thu, 07 Oct 2010) | 9 lines
Ensure editline cleanup occurs when Ctrl-C is pressed at control console.
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
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Review: https://reviewboard.asterisk.org/r/949/
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(closes issue #18001)
Reported by: jamicque
Patches:
20100926__issue18001.diff.txt uploaded by tilghman (license 14)
Tested by: jamicque
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matcher -- as visual candy to be ignored.
Also change the AEL parser to not generate dashes within extensions, as those
dashes would be ignored. Update the AEL tests to match this behavior.
(closes issue #17366)
Reported by: murf
Patches:
20100727__issue17366.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
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r289949 | oej | 2010-10-02 10:50:05 +0200 (Lör, 02 Okt 2010) | 2 lines
Add documentation for undocumented option to AMI action originate
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r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010) | 15 lines
Change RFC2833 DTMF event duration on end to report actual elapsed time.
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
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r289338 | qwell | 2010-09-29 15:56:26 -0500 (Wed, 29 Sep 2010) | 8 lines
Allow a manager originate to succeed on forwarded devices.
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
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r289177 | mnicholson | 2010-09-29 10:03:27 -0500 (Wed, 29 Sep 2010) | 8 lines
Set the caller id on CDRs when it is set on the parent channel.
(closes issue #17569)
Reported by: tbelder
Patches:
17569.diff uploaded by tbelder (license 618)
Tested by: tbelder
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r289094 | bbryant | 2010-09-28 14:10:19 -0400 (Tue, 28 Sep 2010) | 14 lines
Fixes an issue with the Newchannel AMI event during the Masquerading process.
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/
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r288636 | tilghman | 2010-09-23 22:20:24 -0500 (Thu, 23 Sep 2010) | 2 lines
Solaris compatibility fixes
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r288339 | russell | 2010-09-22 11:39:16 -0500 (Wed, 22 Sep 2010) | 11 lines
Fix a 100% CPU consumption problem when setting console=yes in asterisk.conf.
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!
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r288005 | bbryant | 2010-09-21 15:43:46 -0400 (Tue, 21 Sep 2010) | 8 lines
Add a check to fix a rare segmentation fault you'd get if ast_frdup couldn't allocate
memory on the first frame being queued in ast_queue_frame.
(closes issue #17882)
Reported by: seanbright
Tested by: seanbright
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r287933 | tilghman | 2010-09-21 14:07:07 -0500 (Tue, 21 Sep 2010) | 2 lines
Less than zero is an error, not any non-zero value.
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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:
based on bug16057.diff4.txt uploaded by alecdavis (license 585)
Tested by: ramonpeek, davidw, alecdavis
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Merged revisions 287555 via svnmerge from
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r287555 | mnicholson | 2010-09-20 10:48:14 -0500 (Mon, 20 Sep 2010) | 5 lines
Use ast_dynamic_str when processing hint state changes
(related to issue #17928)
Reported by: mdu113
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r287469 | oej | 2010-09-19 17:56:50 +0200 (Sön, 19 Sep 2010) | 7 lines
Make sure we always free variables properly in manager originate.
(closes issue #17891)
reported, solved and tested by oej
Review: https://reviewboard.asterisk.org/r/869/
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r287307 | mnicholson | 2010-09-17 08:34:34 -0500 (Fri, 17 Sep 2010) | 5 lines
Use ast_strdup() instead of ast_strdupa() while processing in ast_hint_state_changed().
(related to issue #17928)
Reported by: mdu113
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r287118 | mnicholson | 2010-09-16 15:04:46 -0500 (Thu, 16 Sep 2010) | 8 lines
Don't limit hint processing in ast_hint_state_changed() to AST_MAX_EXTENSION length strings.
(closes issue #17928)
Reported by: mdu113
Patches:
20100831__issue17928.diff.txt uploaded by tilghman (license 14)
Tested by: mdu113
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r287114 | mnicholson | 2010-09-16 14:52:39 -0500 (Thu, 16 Sep 2010) | 8 lines
Don't stop printing cdr variables if we encounter one with a blank name or value.
(closes issue #17900)
Reported by: under
Patches:
core-show-channel-cdr-fix1.diff uploaded by mnicholson (license 96)
Tested by: mnicholson
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r286679 | mnicholson | 2010-09-14 13:00:01 -0500 (Tue, 14 Sep 2010) | 7 lines
Only drop duplicate answer frames if the channel is bridged.
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)
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r286267 | oej | 2010-09-11 18:59:20 +0200 (Lör, 11 Sep 2010) | 4 lines
Handle error response when we can't make file compatible
Review: https://reviewboard.asterisk.org/r/911/
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r286023 | tilghman | 2010-09-10 13:22:04 -0500 (Fri, 10 Sep 2010) | 2 lines
Missing newline
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r285742 | qwell | 2010-09-09 15:06:31 -0500 (Thu, 09 Sep 2010) | 9 lines
Transmit silence when reading DTMF in ast_readstring.
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)
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pattern ranges and pattern special characters was inconsistent.
(closes issue #16903)
Reported by: Nick_Lewis
Patches:
pbx.c-specificity.patch uploaded by Nick Lewis (license 657)
Tested by: Nick_Lewis
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r280448 | dvossel | 2010-07-29 14:04:23 -0500 (Thu, 29 Jul 2010) | 12 lines
fixes issue with translator frame not getting freed
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
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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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r278982 | tilghman | 2010-07-23 11:43:34 -0500 (Fri, 23 Jul 2010) | 15 lines
Merged revisions 278981 via svnmerge from
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r278981 | tilghman | 2010-07-23 11:42:25 -0500 (Fri, 23 Jul 2010) | 8 lines
Avoid race with consolethread on shutdown (on parallel processors).
(closes issue #17080)
Reported by: sybasesql
Patches:
20100721__issue17080.diff.txt uploaded by tilghman (license 14)
Tested by: sybasesql
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r278708 | jpeeler | 2010-07-22 14:45:30 -0500 (Thu, 22 Jul 2010) | 16 lines
Add method for finding XML doc files for systems that don't support GLOB_BRACE.
In particular, Solaris and perhaps others do not support the above mentioned
GNU extension. In this case the paths are simply expanded without the braces
and the calls to glob are made separately.
Note: I could not explain memory allocation failures that were being reported
from within libxml itself when making calls to glob without using GLOB_NOCHECK.
This is the only reason why that flag is being used.
(closes issue #15402)
Reported by: snuffy
Patches:
bug_xmlpatt-v3.diff uploaded by snuffy (license 35),
modified by me
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r278620 | mmichelson | 2010-07-22 09:58:01 -0500 (Thu, 22 Jul 2010) | 19 lines
Merged revisions 278618 via svnmerge from
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r278618 | mmichelson | 2010-07-22 09:55:04 -0500 (Thu, 22 Jul 2010) | 13 lines
Allow PLC to function properly when channels use SLIN for audio.
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
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r278272 | tilghman | 2010-07-20 17:26:23 -0500 (Tue, 20 Jul 2010) | 11 lines
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r278167 | tilghman | 2010-07-20 15:59:06 -0500 (Tue, 20 Jul 2010) | 4 lines
Do not queue up DTMF frames while a call is on hold.
(Fixes ABE-2110)
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r278024 | tilghman | 2010-07-20 11:50:11 -0500 (Tue, 20 Jul 2010) | 14 lines
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r278023 | tilghman | 2010-07-20 11:37:18 -0500 (Tue, 20 Jul 2010) | 7 lines
Off-by-one error
(closes issue #16506)
Reported by: nik600
Patches:
20100629__issue16506.diff.txt uploaded by tilghman (license 14)
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r277945 | jeang | 2010-07-19 16:07:08 -0500 (Mon, 19 Jul 2010) | 15 lines
Merged revisions 277906 via svnmerge from
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r277906 | jeang | 2010-07-19 15:16:36 -0500 (Mon, 19 Jul 2010) | 7 lines
Avoid trying to pickup a parked extension before the park operation is completed.
A crash could occur if the extension is picked up while the parking extension is
being announced. Testing pu->notquiteyet while searching for a parked extension
resolves this crash.
(ABE-2418)
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r277657 | tringenbach | 2010-07-16 18:23:15 -0500 (Fri, 16 Jul 2010) | 16 lines
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r277625 | tringenbach | 2010-07-16 17:43:39 -0500 (Fri, 16 Jul 2010) | 9 lines
Save and restore AST_FLAG_BRIDGE_HANGUP_DONT on attended transfer.
ast_bridge_call() clears AST_FLAG_BRIDGE_HANGUP_DONT. But during an attended
transfer, ast_bridge_call() is called for a second bridge on the same channel,
and it clears that flag, which still needs to get set for when the original
ast_bridge_call() gets control back and checks it.
Review: https://reviewboard.asterisk.org/r/741
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r277331 | mnicholson | 2010-07-16 13:31:08 -0500 (Fri, 16 Jul 2010) | 15 lines
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r277327 | mnicholson | 2010-07-16 13:30:22 -0500 (Fri, 16 Jul 2010) | 8 lines
Interpret device state AST_DEVICE_UNKNOWN as extension state 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
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r277263 | tilghman | 2010-07-16 13:14:05 -0500 (Fri, 16 Jul 2010) | 12 lines
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r277261 | tilghman | 2010-07-16 13:04:11 -0500 (Fri, 16 Jul 2010) | 5 lines
If variable gotten is not set, will segfault on Solaris.
(closes issue #17636)
Reported by: bklang
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r277250 | mnicholson | 2010-07-16 12:30:39 -0500 (Fri, 16 Jul 2010) | 11 lines
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r277247 | mnicholson | 2010-07-16 12:29:57 -0500 (Fri, 16 Jul 2010) | 4 lines
For pass through DTMF tones, measure the actual duration between the begin and end packets on the wire. If it is detected to be less than AST_MIN_DTMF_DURATION, trigger dtmf emulation.
AST-362
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r277143 | seanbright | 2010-07-16 11:20:40 -0400 (Fri, 16 Jul 2010) | 8 lines
Avoid crashing when installing a duplicate translation path with a lower cost.
(closes issue #17092)
Reported by: moy
Patches:
translate.rev254273.patch uploaded by moy (license 222)
Tested by: moy
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r276653 | jpeeler | 2010-07-15 08:51:11 -0500 (Thu, 15 Jul 2010) | 9 lines
Merged revisions 276652 via svnmerge from
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r276652 | jpeeler | 2010-07-15 08:48:58 -0500 (Thu, 15 Jul 2010) | 2 lines
In a perfect world, the frame source would never be NULL. In the meantime, don't crash when it is.
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r276441 | kpfleming | 2010-07-14 15:15:48 -0500 (Wed, 14 Jul 2010) | 4 lines
Don't try to call an embedded module's backup_globals() function until
after confirming it exists.
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