Dummy channels created by ast_dummy_channel_alloc() should be destoyed by
ast_channel_unref(). Using ast_channel_release() needlessly grabs the
channel container lock and can cause a deadlock as a result.
* Analyzed use of ast_dummy_channel_alloc() and made use
ast_channel_unref() when done with the dummy channel. (Primary reason for
the reported deadlock.)
* Made app_dial.c:dial_exec_full() not call ast_call() holding any channel
locks. Chan_local could not perform deadlock avoidance correctly.
(Potential deadlock exposed by this issue. Secondary reason for the
reported deadlock since the held lock was part of the deadlock chain.)
* Fixed some uses of ast_dummy_channel_alloc() not checking the returned
channel pointer for failure.
* Fixed some potential chan=NULL pointer usage in func_odbc.c. Protected
by testing the bogus_chan value.
* Fixed needlessly clearing a 1024 char auto array when setting the first
char to zero is enough in manager.c:action_getvar().
(closes issue ASTERISK-18613)
Reported by: Thomas Arimont
Patches:
jira_asterisk_18613_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: Thomas Arimont
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@337973 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes an issue where the voicemail duration was being reported
with a duration significantly less than the actual sound file duration.
Voicemails that contained mostly silence were reporting the duration of
only the sound in the file, as opposed to the duration of the file with
the silence. This patch fixes this by having two durations reported in
the __ast_play_and_record family of functions - the sound_duration and the
actual duration of the file. The sound_duration, which is optional, now
reports the duration of the sound in the file, while the actual full duration
of the file is reported in the duration parameter. This allows the voicemail
applications to use the sound_duration for minimum duration checking, while
reporting the full duration to external parties if the voicemail is kept.
(issue ASTERISK-2234)
(closes issue ASTERISK-16981)
Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1443
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@337118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The Dial d and H options break DTMF attended transfer atxferdropcall
option.
1) Party A calls party B.
2) Party B does a DTMF attended transfer to Party C.
If the dialplan uses the Dial d or H options to call Party C then the Dial
application answers the call immediately before initiating the call leg to
Party C. The premature answer causes the transfer code to not invoke the
atxferdropcall=no behavior for a blonde transfer since Party C has
"answered". The transfer code thinks that Party B has "consulted" with
Party C when Party B hangs up and completes the transfer to Party A.
Party A now hears ringback until Party C actually answers.
ASTERISK-13294 Dial d option.
ASTERISK-11067 Dial H option to disconnect before answer.
The referenced issues made Dial answer with the d and H options because
many SIP and ISDN phones cannot send DTMF before the call is connected.
* Made require the dialplan to control when or if the call needs to be
answered to use the Dial application d and H options. (The call is no
longer surprise answered when using the Dial d or H options.)
Review: https://reviewboard.asterisk.org/r/1381/
JIRA AST-623
JIRA AST-666
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336658 65c4cc65-6c06-0410-ace0-fbb531ad65f3
a channel lock must never be held with the queues container lock held.
the deadlock occured on masquerade.
the queues container lock is a relic of the past the old queue module lock.
with ao2 there is no need to hold this lock when dealing with members this
patch removes unneeded locks.
(closes issue ASTERISK-18101)
(closes issue ASTERISK-18487)
Reported by: Paul Rolfe, Jason Legault
Tested by: irroot, Jason Legault, Paul Rolfe
Reviewed by: Matthew Nicholson
Review: https://reviewboard.asterisk.org/r/1402/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336093 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a single channel was dialed and there was media to be forwarded to the
calling channel, the media was written without regard for ringback causing
silence to be heard in some circumstances. This regression was introduced
when the meaning of "single" changed to mean only the number of channels
dialed.
(closes issue ASTERISK-18083)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335341 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a SIP phone uses the dial application and receives a 484 Address
Incomplete response, if overlapped dialing is enabled for SIP, then
the 484 Address Incomplete is forwarded back to the SIP phone and the
HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete
application dialplan logic was automatically triggered; now, explicit
dialplan usage of the application is required.
Additionally, this patch adds a new AST_CONTOL_FRAME type called
AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame,
it is an indication that the dialplan expects more digits back from the
device. If the device supports overlap dialing it should attempt to
notify the device that the dialplan is waiting for more digits; otherwise,
it can handle the frame in a manner appropriate to the channel driver.
(closes issue ASTERISK-17288)
Reported by: Mikael Carlsson
Tested by: Matthew Jordan
Review: https://reviewboard.asterisk.org/r/1416/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335064 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This code change fixes a few issues with the voicemail user override of
emailbody and emailsubject, including escaping the strings, potential memory
leaks, and not overriding the voicemail defaults. Revision 325877 fixed this
for ASTERISK-16795, but did not fix it for ASTERISK-16781. A subsequent
check-in prevented 325877 from being applied to 10. This check-in resolves
both issues, and applies the changes to 1.8, 10, and trunk.
(closes issue ASTERISK-16781)
Reported by: Sebastien Couture
Tested by: mjordan
(closes issue ASTERISK-16795)
Reported by: mdeneen
Tested by: mjordan
Review: https://reviewboard.asterisk.org/r/1374
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333339 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The patch that was committed in the 1.6.x versions of Asterisk for
ASTERISK-15862 actually fixed two issues. One was not applicable to 1.8
but the other is. queue_leak.patch fixes the portion applicable to 1.8.
(closes issue ASTERISK-18265)
Reported by: Fred Schroeder
Patches:
queue_leak.patch (license #5049) patch uploaded by mmichelson
Tested by: Thomas Arimont
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fixed load_realtime_queue() leaking a queue reference when it overwrites
q when processing a realtime queue.
(issue ASTERISK-18265)
* Make join_queue() unreference the queue returned by
load_realtime_queue() when it is done with the pointer. The
load_realtime_queue() returns a reference to the just loaded realtime
queue.
* Fixed queues container reference leak in queues_data_provider_get().
* queue_unref() should not return q that was just unreferenced.
* Made logic in __queues_show() and queues_data_provider_get() when
calling load_realtime_queue() easier to understand.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332874 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This update adds a new AMI event, TestEvent, which is enabled when the TEST_FRAMEWORK compiler flag is defined. It also adds initial usage of this event to app_voicemail. The TestEvent AMI event is used extensively by the voicemail tests in the Asterisk Test Suite.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332817 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
* 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
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
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
If a call to MeetMe includes both the dynamic(D) and always request PIN(P)
options, MeetMe will ask for the PIN two times: once for creating the
conference and once for entering the conference. This behavior was introduced
in rev 311616 when adding the CONFFLAG_ALWAYSPROMPT option to the logic branch
controlling PIN entry for joining a conference.
(closes AST-601)
Review: https://reviewboard.asterisk.org/r/1305/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@328770 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When starting a call with originate, and having the callee channel run Bridge() on pickup, we will double free the dialed_interface_info datastore, causing a crash. Make sure to check if the datastore still exists before trying to free it.
(closes issue ASTERISK-17917)
Reported by: Mark Murawski
Tested by: Mark Murawski
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@328663 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A bug existed such that if a user entered a password with '*', and the extension 'a' did not exist, an invalid mailbox would be created and the user authenticated. The code was changed to prevent this from occurring, and to prevent users from having mailboxes or passwords defined that begin with the '*' character.
(closes issue ASTERISK-17443)
Reported by: Kevin Scott Adams
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1316/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@327852 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This matters only when autoconf fails to detect that weak linking is supported.
External optional dependencies will become optional in both cases, as they are
removed at compile time when not detected. However, runtime-optional modules
are made mandatory when weak linking is not found. This change affects only
the external optional dependencies; previously, they were incorrectly required
when weak linking support was not detected.
Patches:
20110702__issue18062__asterisk_trunk.diff.txt by tilghman (License #5003)
Tested by: iasgoscouk
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@326411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Add ActionID support as documented for the QueueRule AMI action.
* Remove documentation for ActionID with the Queues AMI action. The
output does not follow normal AMI response output and there is no place to
put an ActionID header.
(closes issue AST-602)
Reported by: Vlad Povorozniuc
Patches:
jira_ast_602_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: Vlad Povorozniuc, rmudgett
Review: https://reviewboard.asterisk.org/r/1295/
JIRA SWP-3575
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@325610 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When playing back a prompt to a channel, confbridge neglects to check for
hangup events causing lockup condititions for hangups that occur before
actually joining the conference. This change ensures that the user is removed
from the conference in the event of a premature hangup.
Review: https://reviewboard.asterisk.org/r/1277/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@324305 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
