Commit Graph

23536 Commits

Author SHA1 Message Date
Jonathan Rose
89e84ff75d manager: Make challenge work with allowmultiplelogin=no
Prior to this patch, challenge would yield a multiple logins error if used
without providing the username (which isn't really supposed to be an argument
to challenge) if allowmultiplelogin was set to no because allowmultiplelogin
finds a user with a zero length login name. This check is simply disabled for
the challenge action when the username is empty by this patch.

(closes issue ASTERISK-20677)
Reported by: Vladimir
Patches:
    challenge_action_nomultiplelogin.diff uploaded by Jonathan Rose (license 6182)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376728 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-28 16:47:44 +00:00
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  r376691 | rmudgett | 2012-11-27 18:13:10 -0600 (Tue, 27 Nov 2012) | 39 lines
  
  Fix extension matching with the '-' char.
  
  The '-' char is supposed to be ignored by the dialplan extension matching.
  Unfortunately, it's treatment is not handled consistently throughout the
  extension matching code.
  
  * Made the old exten matching code consistently ignore '-' chars.
  
  * Made the old exten matching code consistently handle case in the
  matching.
  
  * Made ignore empty character sets.
  
  * Fixed ast_extension_cmp() to return -1, 0, or 1 as documented.  The only
  user of it in pbx_lua.c was testing for -1.  It was originally returning
  the strcmp() value for less than which is not usually going to be -1.
  
  * Fix character set sorting if the sets have the same number of characters
  and start with the same character.  Character set [0-9] now sorts before
  [02-9a] as originally intended.
  
  * Updated some extension label and priority already in use warnings to
  also indicate if the extension is aliased.
  
  (closes issue ASTERISK-19205)
  Reported by: Philippe Lindheimer, Birger "WIMPy" Harzenetter
  Tested by: rmudgett
  
  Review: https://reviewboard.asterisk.org/r/2201/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-28 00:19:59 +00:00
Richard Mudgett
95a4a82702 Fix extension matching with the '-' char.
The '-' char is supposed to be ignored by the dialplan extension matching.
Unfortunately, it's treatment is not handled consistently throughout the
extension matching code.

* Made the old exten matching code consistently ignore '-' chars.

* Made the old exten matching code consistently handle case in the
matching.

* Made ignore empty character sets.

* Fixed ast_extension_cmp() to return -1, 0, or 1 as documented.  The only
user of it in pbx_lua.c was testing for -1.  It was originally returning
the strcmp() value for less than which is not usually going to be -1.

* Fix character set sorting if the sets have the same number of characters
and start with the same character.  Character set [0-9] now sorts before
[02-9a] as originally intended.

* Updated some extension label and priority already in use warnings to
also indicate if the extension is aliased.

(closes issue ASTERISK-19205)
Reported by: Philippe Lindheimer, Birger "WIMPy" Harzenetter
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/2201/
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2012-11-28 00:13:10 +00:00
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  r376660 | rmudgett | 2012-11-27 14:39:51 -0600 (Tue, 27 Nov 2012) | 27 lines
  
  Remove unnecessary channel module references.
  
  * Removed call to ast_module_user_hangup_all() in res_config_mysql.c since
  it is effectively a noop.  No channels can attach a reference to that
  module.
  
  * Removed call to ast_module_user_hangup_all() in app_celgenuserevent.c.
  The caller of unload_module() has already called it.
  
  * Removed redundant channel module references in pbx_dundi.c.  The
  registered dialplan function callback dispatchers for the read/read2/write
  callbacks already reference the module before calling.
  
  * pbx_dundi: Moved unregistering CLI commands, DUNDi switch, and dialplan
  functions to the first thing the unload_module() does.  This will reduce
  the chance of new channels using DUNDi services while the module is being
  torn down.
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2012-11-27 21:19:40 +00:00
Richard Mudgett
ebcc4e3da1 Remove unnecessary channel module references.
* Removed call to ast_module_user_hangup_all() in res_config_mysql.c since
it is effectively a noop.  No channels can attach a reference to that
module.

* Removed call to ast_module_user_hangup_all() in app_celgenuserevent.c.
The caller of unload_module() has already called it.

* Removed redundant channel module references in pbx_dundi.c.  The
registered dialplan function callback dispatchers for the read/read2/write
callbacks already reference the module before calling.

* pbx_dundi: Moved unregistering CLI commands, DUNDi switch, and dialplan
functions to the first thing the unload_module() does.  This will reduce
the chance of new channels using DUNDi services while the module is being
torn down.
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2012-11-27 20:39:51 +00:00
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  r376630 | rmudgett | 2012-11-27 11:54:25 -0600 (Tue, 27 Nov 2012) | 13 lines
  
  Made AST_LIST_REMOVE() simpler and use better names.
  
  * Update doxygen of AST_LIST_REMOVE().
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2012-11-27 18:20:18 +00:00
Richard Mudgett
dc653e09bb Made AST_LIST_REMOVE() simpler and use better names.
* Update doxygen of AST_LIST_REMOVE().
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2012-11-27 17:54:25 +00:00
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  r376589 | mjordan | 2012-11-22 18:02:23 -0600 (Thu, 22 Nov 2012) | 29 lines
  
  Re-initialize logmsgs mutex upon logger initialization to prevent lock errors
  
  Similar to the patch that moved the fork earlier in the startup sequence to
  prevent mutex errors in the recursive mutex surrounding the read/write thread
  registration lock, this patch re-initializes the logmsgs mutex.  Part of the
  start up sequence before forking the process into the background includes
  reading asterisk.conf; this has to occur prior to the call to daemon in order
  to read startup parameters.  When reading in a conf file, log statements can
  be generated.  Since this can't be avoided, the mutex instead is
  re-initialized to ensure a reset of any thread tracking information.
  
  This patch also includes some additional debugging to catch errors when
  locking or unlocking the recursive mutex that surrounds locks when the
  DEBUG_THREADS build option is enabled.  DO_CRASH or THREAD_CRASH will
  cause an abort() if a mutex error is detected.
  
  (issue ASTERISK-19463)
  Reported by: mjordan
  Tesetd by: mjordan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376596 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-23 00:20:55 +00:00
Matthew Jordan
588e90a4d9 Re-initialize logmsgs mutex upon logger initialization to prevent lock errors
Similar to the patch that moved the fork earlier in the startup sequence to
prevent mutex errors in the recursive mutex surrounding the read/write thread
registration lock, this patch re-initializes the logmsgs mutex.  Part of the
start up sequence before forking the process into the background includes
reading asterisk.conf; this has to occur prior to the call to daemon in order
to read startup parameters.  When reading in a conf file, log statements can
be generated.  Since this can't be avoided, the mutex instead is
re-initialized to ensure a reset of any thread tracking information.

This patch also includes some additional debugging to catch errors when
locking or unlocking the recursive mutex that surrounds locks when the
DEBUG_THREADS build option is enabled.  DO_CRASH or THREAD_CRASH will
cause an abort() if a mutex error is detected.

(issue ASTERISK-19463)
Reported by: mjordan
Tesetd by: mjordan
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2012-11-23 00:02:23 +00:00
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  r376575 | rmudgett | 2012-11-21 12:33:16 -0600 (Wed, 21 Nov 2012) | 20 lines
  
  Add red-black tree container type to astobj2.
  
  * Add red-black tree container type.
  
  * Add CLI command "astobj2 container dump <name>"
  
  * Added ao2_container_dump() so the container could be dumped by other
  modules for debugging purposes.
  
  * Changed ao2_container_stats() so it can be used by other modules like
  ao2_container_check() for debugging purposes.
  
  * Updated the unit tests to check red-black tree containers.
  
  (closes issue ASTERISK-19970)
  Reported by: rmudgett
  Tested by: rmudgett
  
  Review: https://reviewboard.asterisk.org/r/2110/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-21 19:20:22 +00:00
Richard Mudgett
4ccf2c7aa5 Add red-black tree container type to astobj2.
* Add red-black tree container type.

* Add CLI command "astobj2 container dump <name>"

* Added ao2_container_dump() so the container could be dumped by other
modules for debugging purposes.

* Changed ao2_container_stats() so it can be used by other modules like
ao2_container_check() for debugging purposes.

* Updated the unit tests to check red-black tree containers.

(closes issue ASTERISK-19970)
Reported by: rmudgett
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/2110/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376575 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-21 18:33:16 +00:00
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  r376562 | dlee | 2012-11-20 16:06:05 -0600 (Tue, 20 Nov 2012) | 8 lines
  
  Added missing newlines to websocket ast_logs.
  
  Without these newlines, log messages just continue tacking onto the same
  line, and do not flush immediately.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376567 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-20 22:19:11 +00:00
David M. Lee
cc01a79463 Added missing newlines to websocket ast_logs.
Without these newlines, log messages just continue tacking onto the same
line, and do not flush immediately.
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2012-11-20 22:06:05 +00:00
Mark Michelson
86f1378ea0 Remove extra ast_backtrace() call I had in while debugging.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-20 19:23:55 +00:00
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  r376551 | mmichelson | 2012-11-20 13:09:37 -0600 (Tue, 20 Nov 2012) | 20 lines
  
  Add "Require: timer" to 200 OK responses when appropriate.
  
  The method by which the Require header is added to 200 responses is
  inspired by the method that Olle Johansson uses in his darjeeling-prack
  branch.
  
  (closes issue ASTERISK-20570)
  Reported by Matt Jordan, at the behest of Olle Johansson
  
  Review: https://reviewboard.asterisk.org/r/2172
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2012-11-20 19:19:31 +00:00
Mark Michelson
b37ab7e673 Add "Require: timer" to 200 OK responses when appropriate.
The method by which the Require header is added to 200 responses is
inspired by the method that Olle Johansson uses in his darjeeling-prack
branch.

(closes issue ASTERISK-20570)
Reported by Matt Jordan, at the behest of Olle Johansson

Review: https://reviewboard.asterisk.org/r/2172
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2012-11-20 19:09:37 +00:00
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  r376541 | alecdavis | 2012-11-20 11:39:11 -0600 (Tue, 20 Nov 2012) | 19 lines
  
  Reduce CLI spam of "Extension Changed" device state messages.
  
  Asterisk 11 follows RFC3265 that states that after every subscribe or resubscribe a notify should be sent.
  Thus the console if filled continuously with the following after every subscribe;
    == Extension Changed 8512[phones] new state IDLE for Notify User cisco1
   
  In Asterisk 1.8 only changes would be sent. Thus only when a device state changed was anything emitted to the console.
  
  fix:
  Only print to console when device state isn't forced.
  
  (closes ASTERISK-20706)
  Reported by: alecdavis
  Tested by: alecdavis
  alecdavis (license 585)
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2012-11-20 18:19:37 +00:00
Alec L Davis
316fbb083c Reduce CLI spam of "Extension Changed" device state messages.
Asterisk 11 follows RFC3265 that states that after every subscribe or resubscribe a notify should be sent.
Thus the console if filled continuously with the following after every subscribe;
  == Extension Changed 8512[phones] new state IDLE for Notify User cisco1
 
In Asterisk 1.8 only changes would be sent. Thus only when a device state changed was anything emitted to the console.

fix:
Only print to console when device state isn't forced.

(closes issue ASTERISK-20706)
Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)
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2012-11-20 17:39:11 +00:00
Mark Michelson
e2196d7981 Get rid of trailing whitespace.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376500 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-19 22:34:27 +00:00
Mark Michelson
f4328e109d Reorganize code and change behavior of ast_taskprocessor_execute() when taskprocessor is shutting down.
Moved code around to be easier to follow.

ast_taskprocessor_execute() will now return 0 if the taskprocessor is being shut down.



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2012-11-19 21:31:32 +00:00
Mark Michelson
ff06346ea2 Add doxygen to taskprocessor tests.
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2012-11-19 20:35:39 +00:00
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  r376472 | wdoekes | 2012-11-19 14:03:56 -0600 (Mon, 19 Nov 2012) | 17 lines
  
  Fix most leftover non-opaque ast_str uses.
  
  Instead of calling str->str, one should use ast_str_buffer(str). Same
  goes for str->used as ast_str_strlen(str) and str->len as
  ast_str_size(str).
  
  Review: https://reviewboard.asterisk.org/r/2198
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2012-11-19 20:20:36 +00:00
Walter Doekes
907050d41b Fix most leftover non-opaque ast_str uses.
Instead of calling str->str, one should use ast_str_buffer(str). Same
goes for str->used as ast_str_strlen(str) and str->len as
ast_str_size(str).

Review: https://reviewboard.asterisk.org/r/2198
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2012-11-19 20:03:56 +00:00
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  r376457 | mjordan | 2012-11-18 20:14:54 -0600 (Sun, 18 Nov 2012) | 7 lines
  
  Fix uninitialized in this function error
  
  With some versions of gcc, n_buckets will be flagged as being uninitialized
  before use.  While its technically impossible (since the switch statement,
  even without a default, accounts for all possibilities), we'll initialize the
  variable to 0 anyway.
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2012-11-19 02:19:08 +00:00
Matthew Jordan
7cf42e90b6 Fix uninitialized in this function error
With some versions of gcc, n_buckets will be flagged as being uninitialized
before use.  While its technically impossible (since the switch statement,
even without a default, accounts for all possibilities), we'll initialize the
variable to 0 anyway.


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2012-11-19 02:14:54 +00:00
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  r376447 | mjordan | 2012-11-18 14:27:45 -0600 (Sun, 18 Nov 2012) | 55 lines
  
  Reorder startup sequence to prevent lockups when process is sent to background
  
  Although it is very rare and timing dependent, the potential exists for the
  call to 'daemon' to cause what appears to be a deadlock in Asterisk during
  startup.  This can occur when a recursive mutex is obtained prior to the
  daemon call executing.  Since daemon uses fork to send the process into the
  background, any threading primitives are unsafe to re-use after the call.
  Implementations of pthread recursive mutexes are highly likely to store the
  thread identifier of the thread that previously obtained the mutex.  If
  the mutex was locked prior to the fork, a subsequent unlock operation will
  potentially fail as the thread identifier is no longer valid.  Since the
  mutex is still locked, all subsequent attempts to grab the mutex by other
  threads will block.
  
  This behavior exhibited itself most often when DEBUG_THREADS was enabled, as
  this compile time option surrounds the mutexes in Asterisk with another
  recursive mutex that protects the storage of thread related information.  This
  made it much more likely that a recursive mutex would be obtained prior to
  daemon and unlocked after the call.
  
  This patch does the following:
  a) It backports a patch from Asterisk 11 that prevents the spawning of the
     localtime monitoring thread.  This thread is now spawned after Asterisk has
     fully booted.
  b) It re-orders the startup sequence to call daemon earlier during Asterisk
     startup.  This limits the potential of threading primitives being accessed
     by initialization calls before daemon is called.
  c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called.
     Developers should send error messages directly to stderr prior to daemon,
     as calls to ast_log may access recursive mutexes that store thread related
     information.
  d) It reorganizes when thread local storage is created for storing lock
     information during the creation of threads.  Prior to this patch, the
     read/write lock protecting the list of threads in ast_register_thread would
     utilize the lock in the thread local storage prior to it being initialized;
     this patch prevents that.
  
  On a very related note, this patch will *greatly* improve the stability of the
  Asterisk Test Suite.
  
  Review: https://reviewboard.asterisk.org/r/2197
  
  (closes issue ASTERISK-19463)
  Reported by: mjordan
  Tested by: mjordan
  ........
  
  Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
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................


git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376453 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 21:19:45 +00:00
Matthew Jordan
530ce21b5d Reorder startup sequence to prevent lockups when process is sent to background
Although it is very rare and timing dependent, the potential exists for the
call to 'daemon' to cause what appears to be a deadlock in Asterisk during
startup.  This can occur when a recursive mutex is obtained prior to the
daemon call executing.  Since daemon uses fork to send the process into the
background, any threading primitives are unsafe to re-use after the call.
Implementations of pthread recursive mutexes are highly likely to store the
thread identifier of the thread that previously obtained the mutex.  If
the mutex was locked prior to the fork, a subsequent unlock operation will
potentially fail as the thread identifier is no longer valid.  Since the
mutex is still locked, all subsequent attempts to grab the mutex by other
threads will block.

This behavior exhibited itself most often when DEBUG_THREADS was enabled, as
this compile time option surrounds the mutexes in Asterisk with another
recursive mutex that protects the storage of thread related information.  This
made it much more likely that a recursive mutex would be obtained prior to
daemon and unlocked after the call.

This patch does the following:
a) It backports a patch from Asterisk 11 that prevents the spawning of the
   localtime monitoring thread.  This thread is now spawned after Asterisk has
   fully booted.
b) It re-orders the startup sequence to call daemon earlier during Asterisk
   startup.  This limits the potential of threading primitives being accessed
   by initialization calls before daemon is called.
c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called.
   Developers should send error messages directly to stderr prior to daemon,
   as calls to ast_log may access recursive mutexes that store thread related
   information.
d) It reorganizes when thread local storage is created for storing lock
   information during the creation of threads.  Prior to this patch, the
   read/write lock protecting the list of threads in ast_register_thread would
   utilize the lock in the thread local storage prior to it being initialized;
   this patch prevents that.

On a very related note, this patch will *greatly* improve the stability of the
Asterisk Test Suite.

Review: https://reviewboard.asterisk.org/r/2197

(closes issue ASTERISK-19463)
Reported by: mjordan
Tested by: mjordan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
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  r376416 | mjordan | 2012-11-18 08:31:32 -0600 (Sun, 18 Nov 2012) | 13 lines
  
  Add a test event that reports changes in ConfBridge state
  
  This patch adds a test event to ConfBridge that reports transitions between
  states in ConfBridge.  This is used by tests in the Asterisk Test Suite
  that verify state changes based on the entering/leaving of conference
  participants.
  ........
  
  Merged revisions 376414 from http://svn.asterisk.org/svn/asterisk/branches/10
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376422 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 15:20:26 +00:00
Matthew Jordan
99af7789dd Add a test event that reports changes in ConfBridge state
This patch adds a test event to ConfBridge that reports transitions between
states in ConfBridge.  This is used by tests in the Asterisk Test Suite
that verify state changes based on the entering/leaving of conference
participants.
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376416 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 14:31:32 +00:00
Mark Michelson
623d741ba1 Restructure taskprocessor load test to test integrity of task execution.
This throws a random number to each task and stores a copy locally. After
all tasks have executed, the data is checked to be sure tasks executed
in the correct order.



git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-17 22:27:39 +00:00
Mark Michelson
da0e2169de Add a second default taskprocessor test that executes 20000 tasks.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376412 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-17 21:55:42 +00:00
Mark Michelson
f209bc6f9d Change the default taskprocessor test so it will never hang forever.
Changes the ast_cond_wait() to an ast_cond_timedwait() so that
if there is an issue, we'll never wait forever for the task to
finish execution.



git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-17 21:48:59 +00:00
Mark Michelson
ec68a15619 Test that shutdown callback is called when expected.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376383 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-16 05:00:42 +00:00
Mark Michelson
2b36cbe2d5 Change the write-up on taskprocessors to reflect the new design.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376382 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-16 04:44:12 +00:00
Mark Michelson
12de4198b8 Add a shutdown callback to taskprocessor listeners.
This helps account for the fact that it is unknown just
how many references may exist for a given taskprocessor
listener, so simply unreffing it from the taskprocessor
shutdown function is not enough to convey the gravity
of the situation.

By putting in a shutdown callback, it now becomes clear
to the listener not to try to do any further operations
on the taskprocessor.



git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376381 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-16 04:33:53 +00:00
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................
  r376341 | dlee | 2012-11-15 18:08:00 -0600 (Thu, 15 Nov 2012) | 34 lines
  
  Migrate hashtest/hashtest2 to be unit tests.
  
  Both hashtest and hashtest2 are manual testing apps that thrash hash
  tables (hashtab and ao2 containers, respectively), by spinning up
  several threads that randomly insert, delete, lookup and iterate over
  the hash table. If the app doesn't crash, the hash table probably passes
  the test. Those utils are not a part of the typical Asterisk build, so
  they do not usually get compiled. This all makes them less that useful.
  
  This patch removes those manual test programs and replaces them with
  Asterisk unit test modules (test_{hashtab,astobj2}_thrash.so). It also
  attempts to make the tests more deterministic.
  
  * Rather than spinning up some number of threads that operate on the
    hash table randomly, spin up four threads that concurrenly add,
    remove, lookup and iterate over the hash table.
  * Each thread checks the state of the hash table both during and after
    execution, and indicates a test failure if things are not as expected.
  * Each thread times out after 60 seconds to prevent deadlocking the unit
    test run.
  
  (closes issue ASTERISK-20505)
  Reported by: Matt Jordan
  Review: https://reviewboard.asterisk.org/r/2189/
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  r376344 | dlee | 2012-11-15 18:14:00 -0600 (Thu, 15 Nov 2012) | 1 line
  
  Somehow I put in svn-1.6 merge information. Oops.
................
  r376345 | dlee | 2012-11-15 18:15:30 -0600 (Thu, 15 Nov 2012) | 15 lines
  
  Fixed extconf.c breakage introduced in r376306.
  
  To quote wdoekes:
  > Note that I'm not confirming legitimacy of having that file in tree at
  > all. Is anyone using aelparse/conf2ael?
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376352 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-16 00:19:48 +00:00
David M. Lee
8e07045eb0 Fixed extconf.c breakage introduced in r376306.
To quote wdoekes:
> Note that I'm not confirming legitimacy of having that file in tree at
> all. Is anyone using aelparse/conf2ael?
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376345 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-16 00:15:30 +00:00
David M. Lee
00736a5e5d Somehow I put in svn-1.6 merge information. Oops.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-16 00:14:00 +00:00
David M. Lee
752ffc69d4 Migrate hashtest/hashtest2 to be unit tests.
Both hashtest and hashtest2 are manual testing apps that thrash hash
tables (hashtab and ao2 containers, respectively), by spinning up
several threads that randomly insert, delete, lookup and iterate over
the hash table. If the app doesn't crash, the hash table probably passes
the test. Those utils are not a part of the typical Asterisk build, so
they do not usually get compiled. This all makes them less that useful.

This patch removes those manual test programs and replaces them with
Asterisk unit test modules (test_{hashtab,astobj2}_thrash.so). It also
attempts to make the tests more deterministic.

* Rather than spinning up some number of threads that operate on the
  hash table randomly, spin up four threads that concurrenly add,
  remove, lookup and iterate over the hash table.
* Each thread checks the state of the hash table both during and after
  execution, and indicates a test failure if things are not as expected.
* Each thread times out after 60 seconds to prevent deadlocking the unit
  test run.

(closes issue ASTERISK-20505)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2189/
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2012-11-16 00:08:00 +00:00
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................
  r376312 | jrose | 2012-11-15 17:10:13 -0600 (Thu, 15 Nov 2012) | 23 lines
  
  app_meetme: Fix channels lingering when hung up under certain conditions
  
  Channels would get stuck and MeetMe would repeatedly display an Unable
  to write frame to channel error in the conf_run function if hung up
  during certain sound prompts such as during user count announcements.
  This patch fixes that by reintroducing a hangup check in the meetme's
  main loop (also in conf_run).
  
  (closes issue ASTERISK-20486)
  Reported by: Michael Cargile
  Review: https://reviewboard.asterisk.org/r/2187/
  Patches:
      meetme_hangup_patch_ASTERISK-20486_v3.diff uploaded by Jonathan Rose (license 6182)
  ........
  
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376320 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-15 23:20:11 +00:00
Jonathan Rose
7ec134a8d7 app_meetme: Fix channels lingering when hung up under certain conditions
Channels would get stuck and MeetMe would repeatedly display an Unable
to write frame to channel error in the conf_run function if hung up
during certain sound prompts such as during user count announcements.
This patch fixes that by reintroducing a hangup check in the meetme's
main loop (also in conf_run).

(closes issue ASTERISK-20486)
Reported by: Michael Cargile
Review: https://reviewboard.asterisk.org/r/2187/
Patches:
    meetme_hangup_patch_ASTERISK-20486_v3.diff uploaded by Jonathan Rose (license 6182)
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2012-11-15 23:10:13 +00:00
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  r376291 | beagles | 2012-11-15 08:35:01 -0600 (Thu, 15 Nov 2012) | 14 lines
  
  Patch to prevent stopping the active generator when it is not the silence
  generator.
  
  This patch introduces an internal helper function to safely check whether the
  current generator is the one that is expected before deactivating it. The
  current externally accessible ast_channel_stop_generator() function has been
  modified to be implemented in terms of the new function.
  
  (closes issue ASTERISK-19918)
  Reported by: Eduardo Abad
  ........
  
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376296 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-15 15:20:01 +00:00
Brent Eagles
6b07dd60cc Patch to prevent stopping the active generator when it is not the silence
generator.

This patch introduces an internal helper function to safely check whether the
current generator is the one that is expected before deactivating it. The
current externally accessible ast_channel_stop_generator() function has been
modified to be implemented in terms of the new function.

(closes issue ASTERISK-19918)
Reported by: Eduardo Abad
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2012-11-15 14:35:01 +00:00
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  r376282 | newtonr | 2012-11-14 20:29:40 -0600 (Wed, 14 Nov 2012) | 16 lines
  
  (issue ASTERISK-20280)
  (closes issue ASTERISK-20280)
  Reported by: Tomo Takebe
  Tested by: Rusty Newton
  Patches:
      asterisk20280.patch uploaded by Rusty Newton (license 5829)
  ........
  
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376287 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-15 03:19:12 +00:00
Rusty Newton
21a0dadac0 Patch to play correct sound file when a voicemail's urgent status is removed
We were attempting to play "vm-urgent-removed", which didn't exist. Now we play "vm-marked-nonurgent" which exists      
and is the correct sound file. Previous behavior was silence and a warning on the CLI.

(issue ASTERISK-20280)
(closes issue ASTERISK-20280)
Reported by: Tomo Takebe
Tested by: Rusty Newton
Patches:
    asterisk20280.patch uploaded by Rusty Newton (license 5829)
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2012-11-15 02:29:40 +00:00
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  r376235 | rmudgett | 2012-11-14 13:55:39 -0600 (Wed, 14 Nov 2012) | 25 lines
  
  Fix call files when astspooldir is relative.
  
  Future dated call files are ignored when astspooldir is relative to the
  current directory.  The queue_file() assumed that the qdir needed to be
  prepended if the given filename did not start with a '/'.  If astspooldir
  is relative it is not going to start from the root directory obviously so
  it will not start with a '/'.  The filename used in queue_file()
  ultimately results in qdir prepended multiple times.
  
  * Made queue_file() not prepend qdir if the filename contains a '/'.
  
  (closes issue ASTERISK-20593)
  Reported by: James Le Cuirot
  Patches:
        0004-Fix-future-call-files-from-relative-directories.patch (license #6439) patch uploaded by James Le Cuirot
  ........
  
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376242 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-14 20:20:21 +00:00
Richard Mudgett
ee3927be17 Fix call files when astspooldir is relative.
Future dated call files are ignored when astspooldir is relative to the
current directory.  The queue_file() assumed that the qdir needed to be
prepended if the given filename did not start with a '/'.  If astspooldir
is relative it is not going to start from the root directory obviously so
it will not start with a '/'.  The filename used in queue_file()
ultimately results in qdir prepended multiple times.

* Made queue_file() not prepend qdir if the filename contains a '/'.

(closes issue ASTERISK-20593)
Reported by: James Le Cuirot
Patches:
      0004-Fix-future-call-files-from-relative-directories.patch (license #6439) patch uploaded by James Le Cuirot
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2012-11-14 19:55:39 +00:00
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  r376219 | jrose | 2012-11-13 13:42:13 -0600 (Tue, 13 Nov 2012) | 12 lines
  
  chan_sip: Add SubscribeContext field to SIPshowpeer AMI response
  
  The new field is will show up within the response if the requested peer has a
  subscribe context set.
  
  (closes issue ASTERISK-20626)
  Reported by: Jaco Kroon
  Patches:
      asterisk-sip-ami-SubscrContext.patch uploaded by jkroon (license 5671)
          -with modifications by jrose to conform to style guidelines
  Review: https://reviewboard.asterisk.org/r/2195/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376224 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-13 20:19:49 +00:00
Jonathan Rose
e62bab8131 chan_sip: Add SubscribeContext field to SIPshowpeer AMI response
The new field is will show up within the response if the requested peer has a
subscribe context set.

(closes issue ASTERISK-20626)
Reported by: Jaco Kroon
Patches:
    asterisk-sip-ami-SubscrContext.patch uploaded by jkroon (license 5671)
        -with modifications by jrose to conform to style guidelines
Review: https://reviewboard.asterisk.org/r/2195/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376219 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-13 19:42:13 +00:00
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  r376148 | elguero | 2012-11-12 14:18:47 -0600 (Mon, 12 Nov 2012) | 26 lines
  
  Fix Dynamic Hints Variable Substition - Underscore Problem
  
  When adding a dynamic hint, if an extension contains an underscore no variable
  subsitution is being performed.
  
  This patch changes from checking if the extension contains an underscore to
  checking if the extension begins with an underscore.
  
  (closes issue ASTERISK-20639)
  Reported by: Steven T. Wheeler
  Tested by: Steven T. Wheeler, Michael L. Young
  Patches:
    asterisk-20639-dynamic-hint-underscore.diff 
                                       uploaded by Michael L. Young (license 5026)
  
  Review: https://reviewboard.asterisk.org/r/2188/
  ........
  
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  r376169 | file | 2012-11-12 14:46:51 -0600 (Mon, 12 Nov 2012) | 20 lines
  
  Properly check if the "Context" and "Extension" headers are empty in a ShowDialPlan action.
  
  The code which handles the ShowDialPlan action wrongly assumed that a non-NULL return value
  from the function which retrieves headers from an action indicates that the header has a
  value. This is incorrect and the contents must be checked to see if they are blank.
  
  (closes issue ASTERISK-20628)
  Reported by: jkroon
  Patches:
       asterisk-showdialplan-incorrect-error.patch uploaded by jkroon
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  Merged revisions 376166 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  Merged revisions 376167 from http://svn.asterisk.org/svn/asterisk/branches/10
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  Merged revisions 376168 from http://svn.asterisk.org/svn/asterisk/branches/11
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376176 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-12 21:20:14 +00:00