Calling ast_load_realtime() can leak the last list node if the read list
only contains empty variable value items.
* Fixed list filter loop in ast_load_realtime() to delete the list node
immediately instead of the next time through the loop. The next time
through the loop may not happen if the node to delete is the last in the
list.
(issue ASTERISK-18277)
(issue ASTERISK-18265)
Patches:
jira_asterisk_18265_v1.8_config.patch (license #5621) patch uploaded by rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332759 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The FreeBSD netsock2 test has been failing for a while. We were
pasing sa->len to getnameinfo instead of sa_tmp->len.
ASTERISK-18289
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
JIRA ASTERISK-17183
Multi-parkinglot directs calls to wrong parkinglot.
JIRA ASTERISK-17870
Cannot retrieve parked calls.
JIRA ASTERISK-17430
ParkedCall() with no extension should pickup first available call and does not.
JIRA AST-576
Issues with parking lots
* Removed searching for parking lots by extension. Parking lots can only
be found by the parking lot name since parking lot access extensions and
spaces are not guaranteed to be unique.
* Added parking_lot_name option to the Park and ParkedCall applications.
Updated documentation for Park and ParkedCall applications.
* Add parkext_exclusive configuration option to make parking entry
extensions specify which parking lot they access.
(closes issue ASTERISK-17183)
Reported by: David Cabrejos
Tested by: rmudgett, David Cabrejos
(closes issue ASTERISK-17870)
Reported by: Remi Quezada
(closes issue ASTERISK-17430)
Reported by: Philippe Lindheimer
JIRA ASTERISK-17452
Parking_offset not used
JIRA AST-624
'next' setting for findslot does nothing
* Reimplemented since findslot feature option broken by -r114655.
(closes issue ASTERISK-17452)
Reported by: David Woolley
Tested by: rmudgett
JIRA ASTERISK-15792
Dialplan continues execution after transfer to park.
This happens for DTMF attended transfer, DTMF blind transfer, and DTMF
one-touch-parking if the party initiating these features also initiated
the call.
* Fixed the return code from the affected builtin features when parking a
call.
(closes issue ASTERISK-15792)
Reported by: Mat Murdock
Tested by: rmudgett, twilson
JIRA AST-607
The courtesytone is not playing to the expected call when picking up a
parked call.
This is mostly a documentation problem. However, the option is not reset
to the default when features.conf is reloaded.
* Updated features.conf.sample documentation for courtesytone and
parkedplay options.
* Reset the parkedplay option to default when features.conf is reloaded.
JIRA AST-615
AMI Park action followed by features reload results in orphaned channels
in parking lot.
* Reloading features.conf will not touch parking lots that have calls
still parked in them. Reload again at a later time.
Misc additional fixes:
* Added unit test for parking lot dialplan usage checking.
* Made update connected line when a parked call is retrieved from a
parking lot.
* Made retrieved parked call stop ringing or MOH depending upon how the
call was waiting in the parking lot.
* Made CLI "features show" indicate if the parking lot is enabled for use.
* Added PARKINGDYNEXTEN channel variable to allow dynamic parking lots to
specify the parking lot access extension.
* Made AMI ParkedCalls action ParkedCall events have a Parkinglot header.
* Made AMI ParkedCalls action ParkedCallsComplete event have a Total
header.
* Fixed potential deadlock from AMI Park action holding channel locks
while calling masq_park_call().
* Fixed several places where ast_strdupa() were used inside of loops.
(Mostly fixed by refactoring the loop body into its own function.)
* Fixed copy_parkinglot() copying too much from the source parking lot.
Extracted the parking lot configuration settings into struct
parkinglot_cfg.
* Refactored courtesytone playing code to put the channel not playing the
tone in autoservice.
* Fix when pbx-parkingfailed is played that the other channel is put in
autoservice if it exists.
* Fixed parkinglot reference leak in parked_call_exec() error paths.
* Fixed parkinglot_unref() use of parkinglot after it was unreffed.
* Made destroy the struct ast_parkinglot parkings lock when done.
* Refactored the features.conf parking lot configuration code to eliminate
redundancy.
* Fixed feature reload to better protect parking lots.
* Fixed parking lot container reference leak in handle_parkedcalls().
* Fixed the total count in handle_parkedcalls().
Review: https://reviewboard.asterisk.org/r/1358/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332100 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
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
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
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
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
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
When deciding whether Asterisk was allowed to bridge the call away from the
core, chan_sip did not take into account the usage of features on dialed
channels that require monitoring of DTMF on channels utilizing inband DTMF.
This would cause Asterisk to allow the call to be locally or remotely bridged,
preventing access to the data required to detect activations of such features.
(closes 17237)
Review: https://reviewboard.asterisk.org/r/1302/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@328823 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In some cases when starting asterisk with -c and hitting control-c to shutdown, there will be an invalid read and null pointer deref causing a crash.
(closes issue ASTERISK-17927)
Reported by: Mark Murawski
Tested by: Mark Murawski, Kinsey Moore, Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@328593 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The process_output() function calls ast_str_append() and xml_translate() on its
'out' parameter, which is a pointer to an ast_str buffer. If either of these
functions need to reallocate the ast_str so it will have more space, they will
free the existing buffer and allocate a new one, returning the address of the
new one. However, because process_output only receives a pointer to the ast_str,
not a pointer to its caller's variable holding the pointer, if the original
ast_str is freed, the caller will not know, and will continue to use it (and
later attempt to free it).
(reported by jkroon on #asterisk-dev)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@327950 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It is possible for a dialplan backend to not modify the given buffer or ast_str
and still return success. This causes any previous value stored in the buffer
to be used as if the new function call provided it. Some functions also append
to the given buffer assuming it is empty.
The test_substitution unit test has also been modified to detect this problem.
(closes issue ASTERISK-17878)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@327106 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Mostly comment and format changes.
* ast_context_remove_extension_callerid() and ast_add_extension_nolock()
will write lock the found specific context.
* ast_context_find() will now tolerate a NULL name.
* Eliminated some inlined versions of find_context() and
find_context_locked().
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@326985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a cache directory is used, the process is forked and a mv command is executed to move the temporary file to the permanent location. This caused issues with voicemail, where a race condition occurred when the parent expected the file to be in the permanent location prior to the mv command completing. The parent process is now blocked until the mv command completes.
(closes issue ASTERISK-17724)
Reported by: Adiren P.
Tested by: mjordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@326209 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is recommended by the POSIX standard, as well as by the sigaction(2) manpage
for various platforms that we support (e.g. Mac OS X).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@324955 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The idea behind the patch listed below was used, but in a more targeted manner.
There are now address stringification functions for addresses that are meant to
be sent to a remote party. Link-local scope-ids only make sense on the machine
from which they originate and so are stripped in the new functions.
There is also a host sanitization function added to chan_sip which is used
for when peer and dialog tohost fields or sip_registry hostnames are used to
craft a SIP message.
Also added are some basic unit tests for netsock2 address parsing.
(closes issue ASTERISK-17711)
Reported by: ch_djalel
Patches:
asterisk-1.8.3.2-ipv6_ll_scope.patch uploaded by ch_djalel (license 1251)
Review: https://reviewboard.asterisk.org/r/1278/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@324484 65c4cc65-6c06-0410-ace0-fbb531ad65f3
During this function we can not hold the "chan" lock while
doing the masquerade, the explicit goto on the tmp chan, or
the channel alloc. Instead we need to get the channel lock,
store off information about the channel that we need, and
then let the channel lock go for the remainder of the function.
Review: https://reviewboard.asterisk.org/r/1275/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@324364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r323733 | twilson | 2011-06-15 13:13:00 -0500 (Wed, 15 Jun 2011) | 16 lines
Merged revisions 323732 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r323732 | twilson | 2011-06-15 13:06:24 -0500 (Wed, 15 Jun 2011) | 9 lines
Fix DYNAMIC_FEATURES
DYNAMIC_FEATURES were broken by a recent DTMF change. This patch makes
sure that dynamic features are also checked when deciding whether or not
to pass DTMF through or store it for interpreting.
(closes issue ASTERISK-17914)
Reported by: vrban
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................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@323754 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When leaving a voicemail, the MWI message is never sent. The same thing
happens when checking a voicemail and marking it as read.
If you restart Asterisk, everything comes up at that state correctly, but
changes to the messages in voicemail causes the light to not be set
appropriately. Very easy to reproduce.
* Made ast_event_check_subscriber() return TRUE if there are ANY
subscribers to an event type when there are no restricting ie values
passed. This allows an event being queued to be queued.
(closes issue ASTERISK-18002)
Reported by: lmadsen
Tested by: lmadsen, irroot
Patches:
jira_asterisk_18002_v1.8.patch uploaded by rmudgett (License #5621)
(closes issue ASTERISK-18019)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@323669 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r323579 | seanbright | 2011-06-15 11:22:50 -0400 (Wed, 15 Jun 2011) | 32 lines
Merged revisions 323559 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r323559 | seanbright | 2011-06-15 11:15:30 -0400 (Wed, 15 Jun 2011) | 25 lines
Resolve a segfault/bus error when we try to map memory that falls on a page
boundary.
The fix for ASTERISK-15359 was incorrect in that it added 1 to the length of the
mmap'd region. The problem with this is that reading/writing to that extra byte
outside of the bounds of the underlying fd causes a bus error.
The real issue is that we are working with both a FILE * and the raw fd
underneath it and not synchronizing between them. The code that was removed in
ASTERISK-15359 was correct, but we weren't flushing the FILE * before mapping
the fd.
Looking at the manager code in 1.4 reveals that the FILE * in 'struct
mansession' is never used except to create a temporary file that we immediately
fdopen. This means we just need to write a 0 byte to the fd and everything will
just work. The other branches require a call to fflush() which, while not a
guaranteed fix, should reduce the likelihood of a crash.
This all makes sense in my head.
(closes issue ASTERISK-16460)
Reported by: Ravelomanantsoa Hoby (hoby)
Patches:
issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license #5060)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@323608 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Explicity check the last entry in the DB and make sure that we don't iterate
past it. Since there can be no duplicates, this just makes sure that we stop
after matching the last key.
This patch also refactors the code to get away from some code duplication. A
previous patch added many astdb tests and this patch passed them.
Review: https://reviewboard.asterisk.org/r/1259/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@322981 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Deadlock is possible in ast_do_pickup() when holding the target channel
lock and trying to get the chan channel lock. Also, holding the target
lock when calling ast_channel_masquerade() is not a good idea because that
routine does deadlock avoidance.
* Removed the need to hold the target lock after marking the target with a
datastore and getting the connected line data off of the target channel.
* Moved can_pickup() to ast_can_pickup() in features.c. Now all the call
pickup methods use the same basic call pickup availability check.
Review: https://reviewboard.asterisk.org/r/1234/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@322749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit backports a feature in trunk affecting initreqprep so that display name won't
be encoded improperly. Also includes unit tests for the ast_escape_quoted function.
This patch gives 1.8 a much improved outlook in countries which don't use standard
ASCII characters.
(closes issue ASTERISK-16949)
Reported by: Örn Arnarson
Review: https://reviewboard.asterisk.org/r/1235/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@322585 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk attempts to SRV lookup a host name even if the host name is an IP
address. Regression introduced when IPv6 support was added.
* Restored the check in ast_dnsmgr_lookup() to see if the given host name
is an IP address. The IP address could be in either IPv4 or IPv6 formats.
(closes issue ASTERISK-17815)
Reported by: Byron Clark
Tested by: Byron Clark, Richard Mudgett
Patches:
issue19248_v1.8.patch - uploaded by Richard Mudgett (License #5621)
Review: https://reviewboard.asterisk.org/r/1240/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@322425 65c4cc65-6c06-0410-ace0-fbb531ad65f3
