Holding the channel lock when calling update_queue which attempts to lock the
queue lock can cause a deadlock. This deadlock involves the following chain:
1. hold chan lock -> wait queue lock
2. hold queue lock -> wait agent list lock
3. hold agent list lock -> wait chan list lock
4. hold chan list lock -> wait chan lock
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331774 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* The fake event should only be processed by the channel that currently
owns the private and not the associated call waiting or 3-way channel.
JIRA AST-620
JIRA SWP-3616
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331771 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The AMI actions DAHDIHangup and DAHDITransfer have no effect on a DAHDI
channel. These two AMI actions are highly specialized to analog channels
and appear to make the channel behave like a jack port for headsets.
* Made the faked DAHDI event get processed before a normal media stream
read in dahdi_read() instead of trying to trigger an exception read by
setting the AST_FLAG_EXCEPTION flag. Apparently a change was made long
ago that changed how AST_FLAG_EXCEPTION is processed in the core.
Unfortunately, the faked DAHDI events no longer worked when that happened.
* Updated the DAHDI AMI action documentation for the following actions:
DAHDITransfer, DAHDIHangup, DAHDIDialOffhook, DAHDIDNDon, DAHDIDNDoff,
DAHDIShowChannels, and DAHDIRestart.
* Made use sscanf() instead of atoi() for better error checking of the
DAHDIChannel header string.
JIRA AST-620
JIRA SWP-3616
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331714 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Currently, logger only prints an error message to stderr when it fails to open
a logger channel where many users will not see it because the logger lock is
held. The alternative provided by this patch is to log the error to all
attached consoles in the hopes that it will be easier to see. Additionally,
this patch prevents the failed logger channel from being added to the list
where it would silently fail on each call to the Asterisk logger.
(closes issue ASTERISK-16231)
Review: https://reviewboard.asterisk.org/r/1338
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331649 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The return value of popen() was not checked for failure to open.
(closes issue ASTERISK-18109)
JIRA SWP-3633
Reported by: Michael Myles
Tested by: rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331575 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Any SIP notify sent via AMI or CLI leaks a SIP PVT with ref count +2. Removing
the additional ref just before the invite and adding an unref following it
corrects the issue as seen via REF_DEBUG. The unref existed in a distant
revision and it appears as though the wrong ref operation was removed.
(closes issue ASTERISK-18091)
Review: https://reviewboard.asterisk.org/r/1332/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ASTERISK-15863 caused a regression with queue logging. The output of the
queue log is not started until the logger configuration is reloaded.
* Queue log initialization is completely delayed until the first message
is posted to the queue log system. Including the initial opening of the
queue log file.
* Fixed rotate_file() ROTATE strategy to give the file just rotated out to
the configured exec function after rotate. Just like the other strategies.
* Fixed logger reload to always post the queue reload entry instead of
just if there is a queue log file.
* Refactored some code to eliminate some redundancy and to reduce stack
utilization.
(closes issue ASTERISK-17036)
JIRA SWP-2952
Reported by: Juan Carlos Valero
Patches:
jira_asterisk_17036_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett
(closes issue ASTERISK-18208)
Reported by: Christian Pinedo
Review: https://reviewboard.asterisk.org/r/1333/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331461 65c4cc65-6c06-0410-ace0-fbb531ad65f3
An empty string was not being checked for properly causing identification of
the module to be reloaded to fail and return an Error with message
"No such module."
(closes issue AST-616)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331315 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Add some reentrancy protection in pbx.c when creating the contexts_table
hash table.
* Fix inverted test in chan_sip.c conditional code.
* Fix uninitialized variable and use of the wrong variable in chan_iax2.c.
* Fix test of return value in app_parkandannounce.c. Explicitly testing
for -1 is bad if the function does not actually return that value when it
fails.
* Fixup some comments and add some curly braces in features.c.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If the seek value is past the end of file when resuming G.722 MOH, MOH will
cease to function for the duration of the MOH session through all starts and
stops until saved state is cleared. Adjusting the code to guarantee a single
valid read (which is already assumed) fixes the bug.
(closes issue ASTERISK-18077)
Review: https://reviewboard.asterisk.org/r/1328/
Tested-by: Jonathan Rose <jrose@digium.com>
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331038 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When editing a source file in main/editline, the build system does not rebuild
libedit.a and uses the already existing one instead. Adding a PHONY to
CHECK_SUBDIR fixes this problem.
(closes issue ASTERISK-16221)
Patch-by: Walter Doekes
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@330762 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The call pickup feature did not work on DAHDI devices for anything other than
feature codes beginning with * since all feature codes in chan_dahdi were
originally hard-coded to begin with *. This patch is also applied to
chan_dahdi.c to fix this bug with radio modes.
(closes issue AST-621)
Review: https://reviewboard.asterisk.org/r/1336/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@330705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fixes crash in chan_iax2 resulting from an edge case in the
way control frames are queued during calltoken negotiation is complete.
(closes issue ASTERISK-17610)
Reported by: mgrobecker
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@330581 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r330490 | jrose | 2011-08-01 16:08:10 -0500 (Mon, 01 Aug 2011) | 12 lines
Asterisk 18103 - Fix reload crash caused by destroying default parking lot
Default parking lot was being destroyed in reload and was not being rebuilt properly.
This patch keeps features.c reload from destroying the default parking lot in 1.6.2.
Bug was caused by a hasty backport which didn't test reload enough times to catch the
problem.
(Closes Issue ASTERISK-18103)
Reported by: 808blogger
Review: https://reviewboard.asterisk.org/r/1337/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@330491 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When you say the time in spanish and it is 01:00 - 01:59 or 13:00 - 13:59 you
must use female pronunciation "1F". The function "say_date_with_format_es" does
not take this in account.
(closes ASTERISK-15016)
Patch-by: Luis Jimenez
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@330433 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
..........
r330033 | rmudgett | 2011-07-28 11:26:38 -0500 (Thu, 28 Jul 2011) | 15 lines
Datacalls with B410P fail.
Incoming and outgoing call legs of a data call are using different
formats: a-law, u-law. When the call is bridged, the media stream is run
through translation to convert the media formats. The translation is bad
for data calls.
* Make incoming call that does not explicitly specify u-law or a-law use
the DAHDI channel's default law. The outgoing call always uses the
default law from the DAHDI channel.
(closes issue ABE-2800)
Patches:
jira_abe_2800_companding.patch (license #5621) patch uploaded by rmudgett
..........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@330050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Formerly, prepend forwarding would have the user record a message with no useful prompt
and an expectation for the user to push a button on the phone when finished recording.
If a length of silence was detected instead, the recording would be canceled and the user
would re-enter the voicemail forwarding menu. Subsequent time-outs in prepend recording
would also bug out in the sense that they would write over the original message and get
sent to the recipient regardless of whether they timed out or were accepted. This patch
fixes this issue and adds a prompt which will be played after a timeout informing the
user that they needed to press a button. Currently, the sound files that we have are
somewhat inadquate for this, so after the call we simply have Allison say "Please try
again. Then press pound." which actually relies on two separate sound files. Just one
would be more appropriate.
reporter: Vlad Povorozniuc
Review: https://reviewboard.asterisk.org/r/1327/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@329527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Make use buffer accessor function in handle_statechange() rather than
directly accessing the struct member.
* Make use less redundant loop construct for iterating over hints.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@329333 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There are two remaining different deadlocks reported dealing with dialplan
hints.
The deadlock in ASTERISK-17666 is caused by invalid locking order in
ast_remove_hint(). The hints container must be locked before the hint
object.
The deadlock in ASTERISK-17760 is caused by a catch-22 situation in
handle_statechange(). The deadlock is caused by not having the conlock
before calling the watcher callbacks. Unfortunately, having that lock
causes a different deadlock as reported in ASTERISK-16961.
* Fixed ast_remove_hint() locking order.
* Made handle_statechange() no longer call the watcher callbacks holding
any locks that matter.
* Made hint ao2 destructor do the watcher callbacks for extension
deactivation to guarantee that they get called.
* Fixed hint reference leak in ast_add_hint() if the callback container
constructor failed.
* Fixed hint reference leak in complete_core_show_hint() for every hint it
found for CLI tab completion.
* Adjusted locking in ast_merge_contexts_and_delete() for safety.
* Added context_merge_lock to prevent ast_merge_contexts_and_delete() and
handle_statechange() from interfering with each other.
* Fixed ast_change_hint() not taking into account that the extension is
used for the hash key.
(closes issue ASTERISK-17666)
Reported by: irroot
Tested by: irroot
JIRA SWP-3318
(closes issue ASTERISK-17760)
Reported by: Byron Clark
Tested by: irroot
JIRA SWP-3393
Review: https://reviewboard.asterisk.org/r/1313/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@329299 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The PickupChan uses the ampersand as the argument separator.
Was documented as:
PickupChan(channel[,channel2[,...][,options]])
Fixed documentation to:
PickupChan(Technology/Resource[&Technology2/Resource2[&...]][,options])
This is a continuation of ASTERISK-17494 for v1.8 and later.
(closes issue ASTERISK-18144)
Reported by: Erik Smith
Patches:
pickupchan_ducumentation-v2.patch (License #6263) patch uploaded by Erik Smith
Tested by: Erik Smith
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@329199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This appears to be a leftover from when ast_channel was converted to ao2
objects.
Simply removed the extraneous unlock.
(closes issue ASTERISK-17772)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@329144 65c4cc65-6c06-0410-ace0-fbb531ad65f3
