This one involves shrinking the threadpool in such
a way that both idle and active threads are affected.
This test made me re-realize why the zombie state exists,
so I re-added it. We don't want to clog up the control
taskprocessor by waiting on active threads to complete
what they are doing. Instead, we mark them as zombies so
that when they are done, they can clean themselves up
properly.
Without the zombie state available, the new test actually
will deadlock.
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r377384 | kmoore | 2012-12-07 16:08:48 -0600 (Fri, 07 Dec 2012) | 23 lines
codec_dahdi: Fix output of "transcoder show" CLI command.
In r306010 "Asterisk media architecture conversion - no more format
bitfields", the logic for incrementing encoders and decoders when
opening transcoder channels was changed without making the corresponding
change when decrementing encoder / decoder channels. The result being
that when a channel was destroyed, codec_dahdi couldn't properly tell if
it was an encoder or decoder, and the default case is to assume it was a
decoder.
This could result in negative numbers for decoders in use like in:
VOIP6*CLI> transcoder show
2/-2 encoders/decoders of 92 channels are in use.
(closes issue ASTERISK-19921)
Patch-by: Shaun Ruffell
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The new thread creation test fails because Asterisk locks up
while trying to lock a taskprocessor.
While trying to debug that, I found a race condition during taskprocessor
creation where a default taskprocessor listener could try to operate on
a partially started taskprocessor. This was fixed by adding a new callback
to taskprocessor listeners.
Then while testing that change, I found some bugs in the taskprocessor
tests where I was not properly unlocking when done with a lock. Scoped
locks have spoiled me a bit.
I still have not figured out why the threadpool thread creation test
is locking up.
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r377356 | rmudgett | 2012-12-06 18:00:39 -0600 (Thu, 06 Dec 2012) | 30 lines
confbridge: Fix some resource leaks on conference teardown.
* Made destroy_conference_bridge() destroy a missed ast_mutex_t and ast_cond_t.
* Made join_conference_bridge() init the ast_mutex_t's and ast_cond_t so
destroy_conference_bridge() can destroy them unconditionally.
* Made join_conference_bridge() abort if the new conference could not be
added to the conferences container.
* Made leave_conference() discard any post-join actions if
join_conference_bridge() had to abort early.
* Made the join_conference_bridge() diagnostic messages better describe
what happened.
* Renamed leave_conference_bridge() to leave_conference() and made it only
take a conference user pointer. The conference pointer was redundant.
* Made conf_bridge_profile_copy() use struct copy instead of memcpy().
* No need to lock the conference in start_conf_record_thread() since all
of the callers already have it locked.
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r377324 | mjordan | 2012-12-06 08:26:13 -0600 (Thu, 06 Dec 2012) | 13 lines
Fix memory leak in 'manager show event' when command entered incorrectly
When the CLI command 'manager show event' was run incorrectly and its usage
instructions returned, a reference to the event container was leaked. This
would prevent the container from being reclaimed when Asterisk exits. We now
properly decrement the count on the ao2 object using the nifty RAII_VAR macro.
Thanks to Russell for helping me stumble on this, and Terry for writing that
ridiculously helpful macro.
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r377329 | russell | 2012-12-06 09:06:47 -0600 (Thu, 06 Dec 2012) | 7 lines
Add CLI tab completion to 'acl show'.
The 'acl show' CLI command allows you to show the details about a specific
named ACL in acl.conf. This patch adds tab completion to the command.
Review: https://reviewboard.asterisk.org/r/2230/
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r377330 | russell | 2012-12-06 09:13:37 -0600 (Thu, 06 Dec 2012) | 6 lines
Minor code cleanup in named_acl.c.
This patch makes a few little cleanups to named_acl.c. A couple non-public
functions were made static and an opening brace for a function was moved to
its own line, per the coding guidelines.
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r377260 | file | 2012-12-05 10:51:58 -0600 (Wed, 05 Dec 2012) | 25 lines
Fix a SIP request memory leak with TLS connections.
During the TLS re-work in chan_sip some TLS specific code was moved
into a separate function. This function operates on a copy of the
incoming SIP request. This copy was never deinitialized causing a
memory leak for each request processed.
This function is now given a SIP request structure which it can use
to copy the incoming request into. This reduces the amount of memory
allocations done since the internal allocated components are reused
between packets and also ensures the SIP request structure is
deinitialized when the TLS connection is torn down.
(closes issue ASTERISK-20763)
Reported by: deti
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r377263 | jrose | 2012-12-05 11:17:06 -0600 (Wed, 05 Dec 2012) | 21 lines
res_srtp: Fix a crash caused by srtp_dealloc on an already dealloced session
When srtp_create fails, the session may be dealloced or just not alloced. At
the same time though, the session pointer might not be set to NULL in this
process and attempting to srtp_dealloc it again will cause a segfault. This
patch checks for failure of srtp_create and sets the session pointer to NULL
if it fails.
(closes issue ASTERISK-20499)
Reported by: tootai
Review: https://reviewboard.asterisk.org/r/2228/
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r377229 | rmudgett | 2012-12-04 19:11:26 -0600 (Tue, 04 Dec 2012) | 31 lines
confbridge: Fix several small issues.
* Made func_confbridge_helper() allow an empty value when setting options.
You previously could not Set(CONFBRIDGE(user,pin)=) and clear the
configured pin from the dialplan.
* Made func_confbridge_helper() handle its datastore better if multiple
threads attempt to set the first CONFBRIDGE option value on the channel.
* Made the func_confbridge_helper() only output one diagnostic message
concerning the option.
* Made the bridge video_mode able to repeatedly change in the config file
and CONFBRIDGE dialplan function. The video_mode option values are an
enum and not independent of each other.
* Made handle_cli_confbridge_show_bridge_profile() better handle the
video_mode option.
* Simplified datastore handling code in conf_find_user_profile() and
conf_find_bridge_profile().
(closes issue ASTERISK-20655)
Reported by: Birger "WIMPy" Harzenetter
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After giving it some consideration, there's no real
use for zombie threads. Listeners can't really use the
current number of zombie threads as a way of gauging activity,
zombifying threads is just an extra step before they die that
really serves no purpose, and since there's no way to re-animate
zombies, the operation does not need to be around.
I also fixed up some miscellaneous compilation errors that
were lingering from some past revisions.
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Since threadpool shutdown is very strictly controlled,
there is no need to be so precise with reference counts
in queued operations. Since the threadpool shuts down its
own control taskprocessor before doing anything else destructive,
it can be guaranteed that all queued tasks will have a valid
pointer to the pool. This meant that some destructor functions
for helper structs could be removed entirely.
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r377168 | rmudgett | 2012-12-03 17:00:08 -0600 (Mon, 03 Dec 2012) | 21 lines
Cleanup ast_run_atexits() atexits list.
* Convert atexits list to a mutex instead of a rd/wr lock. The lock is
only write locked.
* Move CLI verbose Asterisk ending message to where AMI message is output
in really_quit() to avoid further surprises about using stuff already
shutdown.
(issue ASTERISK-20649)
Reported by: Corey Farrell
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r377018 | oej | 2012-12-03 08:46:02 -0600 (Mon, 03 Dec 2012) | 5 lines
Move functions to AFTER the block of forward declarations of functions.
It was a mess. The first part of chan_sip.c is constants, declarations, structures and stuff,
then forward declarations and then actual code. It's still a mess, but a bit less messy ;-)
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r377022 | file | 2012-12-03 08:56:36 -0600 (Mon, 03 Dec 2012) | 13 lines
Fix an RTP instance reference count leak in chan_motif.
When setting up an RTP instance the RTCP portion of the instance
keeps a reference to the instance itself. In order to release this
reference and stop RTCP the stop API call must be called before
destroying the instance.
(closes issue ASTERISK-20751)
Reported by: joshoa
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r376998 | oej | 2012-12-03 03:35:55 -0600 (Mon, 03 Dec 2012) | 4 lines
Formatting changes
Found a large amount of missing {} in the code before patching in another branch
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r376984 | file | 2012-11-30 18:47:42 -0600 (Fri, 30 Nov 2012) | 10 lines
Tweak extension used for incoming calls received on Motif.
Based on feedback from numerous individuals this patch tweaks incoming calls
to first look for an extension with the name of the endpoint. If no such extension
exists the call will silently fall back to the "s" extension as it previously
did.
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r376918 | mmichelson | 2012-11-30 10:56:53 -0600 (Fri, 30 Nov 2012) | 29 lines
Fix potential crashes during SIP attended transfers.
The principal behind this patch is simple. During a transfer,
we manipulate channels that are owned by a separate thread than
the one we currently are running in, so it makes sense that we
need to grab a reference to the channels so that they cannot
disappear out from under us.
In the wild, crashes were sometimes seen when the transferring
party would hang up the call before the transfer target answered
the call. The most common place to see the crash occur was when
attempting to send a connected line update to the transferer
channel.
(closes issue ASTERISK-20226)
Reported by Jared Smith
Patches:
ASTERISK-20226.patch uploaded by Mark Michelson (License #5049)
Tested by: Jared Smith
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r376922 | seanbright | 2012-11-30 11:08:41 -0600 (Fri, 30 Nov 2012) | 11 lines
Minor spelling fix to the VOLUME documentation.
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r376837 | elguero | 2012-11-29 15:58:41 -0600 (Thu, 29 Nov 2012) | 25 lines
Improve Code Readability And Fix Setting natdetected Flag
For 1.8, 10, 11 and trunk we are are improving the code readability.
For 11 and trunk, auto nat detection was added. The natdetected flag was being
set to 1 when the host address in the VIA header did not specifiy a port. This
patch fixes this by setting the port on the temporary sock address used to
SIP_STANDARD_PORT in order for the sock address comparison to work properly.
(closes issue ASTERISK-20724)
Reported by: Michael L. Young
Patches:
asterisk-20724-set-port-v2.diff uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2206/
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r376820 | pkiefer | 2012-11-29 10:44:42 -0600 (Thu, 29 Nov 2012) | 14 lines
Fix chan_sip websocket payload handling
Websocket by default doesn't return an ast_str for the payload received. When
converting it to an ast_str on chan_sip the last character was being omitted,
because ast_str functions expects that the given length includes the trailing
0x00. payload_len only has the actual string length without counting the
trailing zero.
For most cases this passed unnoticed as most of SIP messages ends with \r\n.
(closes issue ASTERISK-20745)
Reported by: I?\195?\177aki Baz Castillo
Review: https://reviewboard.asterisk.org/r/2219/
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r376821 | dlee | 2012-11-29 11:16:50 -0600 (Thu, 29 Nov 2012) | 5 lines
Fixed ast_random's comment about locking.
The original comment was separated from the code at some point, and didn't
reflect the use of libc's other than glibc for Linux.
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Mark Michelson
Automerge script