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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376457 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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potentially fail as the thread identifier is no longer valid.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
........
Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
........
Merged revisions 376306 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 376315 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 376339 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376341 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
........
Merged revisions 376217 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376291 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
........
Merged revisions 376232 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 376233 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 376234 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376235 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
With ICE support enabled in chan_sip and a large number of interfaces on the system it was
possible for the produced SDP to be truncated due to some fixed size buffers. These buffers
have now been changed so they will dynamically grow as needed.
ICE support is now also enabled by default in res_rtp_asterisk to provide a smoother experience
for chan_motif users where it is required. To maintain the previous behavior in chan_sip it is
no longer enabled by default there.
(closes issue ASTERISK-20643)
Reported by: coopvr
........
Merged revisions 376130 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376131 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Makes malloc() behave like calloc(). It will return a memory block
filled with 0x55. A nonzero value.
* Makes free() fill the released memory block and boundary fence's with
0xdeaddead. Any pointer use after free is going to have a pointer
pointing to 0xdeaddead. The 0xdeaddead pointer is usually an invalid
memory address so a crash is expected.
* Puts the freed memory block into a circular array so it is not reused
immediately.
* When the circular array rotates out a memory block to the heap it checks
that the memory has not been altered from 0xdeaddead.
* Made the astmm_log message wording better.
* Made crash if the DO_CRASH menuselect option is enabled and something is
found.
* Fixed a potential alignment issue on 64 bit systems.
struct ast_region.data[] should now be aligned correctly for all
platforms.
* Extracted region_check_fences() from __ast_free_region() and
handle_memory_show().
* Updated handle_memory_show() CLI usage help.
Review: https://reviewboard.asterisk.org/r/2182/
........
Merged revisions 376029 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 376030 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 376048 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376049 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r375993 | mmichelson | 2012-11-07 11:01:13 -0600 (Wed, 07 Nov 2012) | 30 lines
Fix misuses of timeouts throughout the code.
Prior to this change, a common method for determining if a timeout
was reached was to call a function such as ast_waitfor_n() and inspect
the out parameter that told how many milliseconds were left, then use
that as the input to ast_waitfor_n() on the next go-around.
The problem with this is that in some cases, submillisecond timeouts
can occur, resulting in the out parameter not decreasing any. When this
happens thousands of times, the result is that the timeout takes much
longer than intended to be reached. As an example, I had a situation where
a 3 second timeout took multiple days to finally end since most wakeups
from ast_waitfor_n() were under a millisecond.
This patch seeks to fix this pattern throughout the code. Now we log the
time when an operation began and find the difference in wall clock time
between now and when the event started. This means that sub-millisecond timeouts
now cannot play havoc when trying to determine if something has timed out.
Part of this fix also includes changing the function ast_waitfor() so that it
is possible for it to return less than zero when a negative timeout is given
to it. This makes it actually possible to detect errors in ast_waitfor() when
there is no timeout.
(closes issue ASTERISK-20414)
reported by David M. Lee
Review: https://reviewboard.asterisk.org/r/2135/
........
r375994 | mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) | 3 lines
Remove some debugging that accidentally made it in the last commit.
........
Merged revisions 375993-375994 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 375995 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 376014 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376015 65c4cc65-6c06-0410-ace0-fbb531ad65f3
An issue was reported on the mailing list where calling would result in an "Incomplete
ICE-UDP candidate received on session" error message. This is the result of the ICE-UDP
candidate code not placing a "network" attribute within the candidates. This is now done.
To increase compatibility though I have removed the requirement for the "network" attribute
to exist within ICE-UDP candidates that are received since we don't actually require the
value.
Reported on the mailing list by Jean-Denis Girard.
........
Merged revisions 375925 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375926 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Currently, if an acknowledgement of a timer fails Asterisk will not realize
that a serious error occurred and will continue attempting to use the timer's
file descriptor. This can lead to situations where errors stream to the
CLI/log file. This consumes significant resources, masks the actual problem
that occurred (whatever caused the timer to fail in the first place), and
can leave channels in odd states.
This patch propagates the errors in the timing resource modules up through
the timer core, and makes users of these timers handle acknowledgement
failures. It also adds some defensive coding around the use of timers
to prevent using bad file descriptors in off nominal code paths.
Note that the patch created by the issue reporter was modified slightly for
this commit and backported to 1.8, as it was originally written for
Asterisk 10.
Review: https://reviewboard.asterisk.org/r/2178/
(issue ASTERISK-20032)
Reported by: Jeremiah Gowdy
patches:
jgowdy-timerfd-6-22-2012.diff uploaded by Jeremiah Gowdy (license 6358)
........
Merged revisions 375893 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 375894 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 375895 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375896 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Manager's tcp/tls objects have a periodic function that purge old manager
sessions periodically. During shutdown, the underlying container holding
those sessions can be disposed of and set to NULL before the tcp/tls periodic
function is stopped. If the periodic function fires, it will attempt to
iterate over a NULL container.
This patch checks for whether or not the sessions container exists before
attempting to purge sessions out of it. If the sessions container is NULL,
we simply return.
Note that this error was also caught by the Asterisk Test Suite.
........
Merged revisions 375800 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 375801 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 375802 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375803 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The AstDB uses prepared SQLite3 statements to retrieve data from the SQLite3
database. These statements should be finalized during Asterisk shutdown so
that the SQLite3 database can be properly closed. Failure to finalize the
statements results in a memory leak and a failure when closing the database.
This patch fixes those issues by ensuring that all prepared statements are
properly finalized at shutdown.
(closes issue ASTERISK-20647)
Reported by: Corey Farrell
patches:
astdb-sqlite3_close.patch uploaded by Corey Farrell (license 5909)
........
Merged revisions 375761 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 375763 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375770 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes two memory leaks:
1) When building XML documentation items, the 'name' attribute was extracted
from XML elements but not properly freed after being copied into the item
being built.
2) When unloading XML documentation, the doctree container objects were not
properly freed.
This patch corrects these memory leaks. Note that this patch was modified
slightly for this commmit, as the case where the 'name' attribute doesn't
exist also wasn't handled in the item construction. This patch also checks
for that attribute not existing.
(closes issue ASTERISK-20648)
Reported by: Corey Farrell
Tested by: mjordan
patches:
xmldoc-memory_leak.patch uploaded by Corey Farrell (license 5909)
........
Merged revisions 375756 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The Asterisk Test Suite caught an error condition where a scheduled CDR batch
write can be deleted twice if two channels attempt to post their CDRs at the
same time. The batch CDR mutex is locked while the CDRs are appended to the
current batch list; however, it is unlocked prior to actually scheduling the
CDR write. As such, two threads can attempt to remove the currently scheduled
batch write at the same time, resulting in an assertion error.
This patch extends the time that the mutex is locked to encompass actually
scheduling the write. This prevents two threads from unscheduling the
currently scheduled write at the same time.
........
Merged revisions 375727 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 375728 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 375729 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375730 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r375519 | rmudgett | 2012-10-30 16:06:15 -0500 (Tue, 30 Oct 2012) | 11 lines
chan_misdn: Timer primitives must be handled first.
The frm->addr is a different "address space" than the stack/instance
address of other Lx primitives. The test for B channel instance address
could fail.
Patches:
patch01_timers.diff (license #6372) patch uploaded by Guenther Kelleter
JIRA ABE-2888
........
r375520 | rmudgett | 2012-10-30 16:14:58 -0500 (Tue, 30 Oct 2012) | 10 lines
chan_misdn: Free memory in error paths and other memory leaks.
The one line commented with BUG is not easily fixable because there is no
de-init function one can call.
Patches:
patch02_memory.diff (license #6372) patch uploaded by Guenther Kelleter
JIRA ABE-2888
........
r375521 | rmudgett | 2012-10-30 16:38:41 -0500 (Tue, 30 Oct 2012) | 14 lines
chan_misdn: ISDN NT L2 de-establish/establish
* An NT-PTMP cannot de/establish L2 since it doesn't know the TEIs.
* On NT-PTP L2 is started when L1 is finally active in handle_l1.
* L2 deactivation logging cleanup.
* L2 aggregate link status is unknown for NT-PTMP, show as "UNKN".
* Removed unused functions and code for L2 handling.
Patches:
patch03_L2estab.diff (license #6372) patch uploaded by Guenther Kelleter
Modified
JIRA ABE-2888
........
r375522 | rmudgett | 2012-10-30 16:56:14 -0500 (Tue, 30 Oct 2012) | 22 lines
chan_misdn: Fix broken upper_id/lower_id usage.
Sending PH prim via lower_id layer (3 or 1) simply does not work. For TE
(3) it returns an error (len=-6) which is not evaluated by handle_l1(), so
the L1 layer status ends up wrong. Instead PH must be sent via L4, only
then does it reach L1 without an error message.
And NT PH prims only reach L1 when they are sent to layer 2 id.
--> use upper_id to send PH primitives.
* Check for errors in PH_(DE)ACTIVATE | CONFIRM.
* Debug messages are improved.
* The lower_id is now not used for anything, except: Why is lower_id layer
deleted when it wasn't created? I removed this code since it looks very
wrong.
Patches:
patch04_l1activation.diff (license #6372) patch uploaded by Guenther Kelleter
JIRA ABE-2888
........
r375523 | rmudgett | 2012-10-30 17:29:15 -0500 (Tue, 30 Oct 2012) | 31 lines
chan_misdn: Fix loss of B channels if L1 is down.
If you make 2 calls out an NT PTMP port which is not connected to any
phone, the B channel associated with that call becomes unusable until
Asterisk is restarted.
The problem is the EVENT_SETUP is queued when L1 is not up in
misdn_lib_send_event(). If L1 cannot be activated the event won't be
dequeued. It gets even worse when the call is hung up. The queued
EVENT_SETUP will be overwritten by an EVENT_DISCONNECT. The reserved B
channel then will never be freed. If later someone connects a phone to
the port, L1 will eventually activate and the queued EVENT_DISCONNECT is
sent down the stack. However, it is ignored because it is the wrong call
state.
The real fix would be that activation and queueing for a new SETUP is done
by the NT stack. But since it doesn't, the workaround must be removed
because it doesn't always work.
Fix: The event is no longer queued but immediately sent to the stack. If
L1 cannot be activated, the L3 state machine that was started by the
EVENT_SETUP will do its work, i.e. a timeout will release the B channel
properly. The SETUP possibly cannot be sent the first time but is resent
by T303 in case L1 could be activated.
Patches:
patch05_bchan-loss.diff (license #6372) patch uploaded by Guenther Kelleter
Modified
JIRA ABE-2888
........
r375524 | rmudgett | 2012-10-30 18:26:05 -0500 (Tue, 30 Oct 2012) | 13 lines
chan_misdn: Remove some calls to exit().
Try proper cleanup when something goes wrong in misdn_lib_init().
Especially do not call exit()!
* Fix memory leak because stack_destroy() does not free the stack struct.
Patches:
patch06_cleanup-init.diff (license #6372) patch uploaded by Guenther Kelleter
Modified
JIRA ABE-2888
........
Merged revisions 375519-375524 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
........
Merged revisions 375625 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 375626 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 375627 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375628 65c4cc65-6c06-0410-ace0-fbb531ad65f3
While looking at some debug logs, I noticed that it was being reported that the
SDP origin line was unsupported or failed. Upon looking into this on my local
machine, I found that I too was getting this debug message yet everything seemed
to be getting processed properly. What was discovered is, that, the variable to
determine what is displayed in the debug message for the SDP line that was
processed, was not being set for the origin line when the result was successful.
This patch fixes this and was tested on local machine.
........
Merged revisions 375594 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 375601 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 375613 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375614 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A regression was introduced in chan_sip by changes to sip reload introduced by
r349097. That patch moved peer purging from the beginning of the reload to
after the general configuration was finished. This patch fixes that by undoing
the repositioning of the original peer purging code and using a similar
function after performing general configuration that purges only autocreated
peers that were created when persist mode isn't enabled.
(closes issue ASTERISK-20611)
Reported by: Alisher
Review: https://reviewboard.asterisk.org/r/2171/
........
Merged revisions 375575 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375576 65c4cc65-6c06-0410-ace0-fbb531ad65f3
On some systems the optional API support uses the GCC compiler attribute "weakref" to provide its
functionality. This code changes the function names and prefixes "__" to the front. The
res_http_websocket exports file did not take this into account, thereby not allowing those functions
to be global and ultimately found.
(closes issue ASTERISK-20631)
Reported by: danjenkins
........
Merged revisions 375559 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Unlike all other calendar modules, res_calendar_ews fails to extract the Body
information for a calendar item. This is due, in part, to a quirk in the
schema in the XML - not only does a CalendarItem contain a Body element, but
the CalendarItem exists as a descendant of a different Body element. The neon
parser was erroneously skipping all Body elements.
This patch fixes that by bypassing Body elements that are not a child of
CalendarItem, and parsing the Body element out if it is a child.
Note that the original patch by Terry Wilson only needed slight modifications
to make it properly pull the Body information out; as such, while I've linked
to the patch that I uploaded for Dmitry, I've attributed the patch to Terry.
(closes issue ASTERISK-19738)
Reported by: Dmitry Burilov
Tested by: Dmitry Burilov
patches:
calendar_ews_body_2012_10_29.diff uploaded by Terry Wilson (license 6283)
........
Merged revisions 375528 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 375531 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 375532 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375533 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When confbridge was changed to handle conference status with a state machine in
r374658. The function responsible for starting recording for a conference was
refactored with the function actually responsible for launching the recording
thread being split into a function with another name. The old function name was
still used for manually started recordings through AMI or CLI. This patch fixes
that by switching which function is used to start recording the conference.
(closes issue ASTERISK-20601)
Reported by: Vilius
Patches:
confbridge_mixmonitor.diff uploaded by Jonathan Rose (license 6182)
........
Merged revisions 375470 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 375471 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375472 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a "sip reload" is issued for a SIP peer, then his
IP address will be cleared, thus resulting in forgetting the
public IP address. Asterisk will then attempt to route SIP
traffic to the private IP address.
The fix here is to make "sip reload" ignore realtime peers
when "host = dynamic" is spotted. Realtime peers can now only
have their IP address reset if they have gone from being not
dynamic to being dynamic.
(closes issue ASTERISK-18203)
reported by daren ferreira
(closes issue ASTERISK-20572)
reported by JoshE
Patches:
fix_nat_realtime.diff uploaded by JoshE (license #6075)
........
Merged revisions 375415 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 375417 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 375437 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375443 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a caller enters a queue and no queue member answers the call, the current
behaviour can be a little odd depending on the paused status of the queue
members. If any queue member is paused, but not all, the CDR disposition
will be BUSY. If all queue members are paused, then the CDR disposition is
based instead on the disposition of the call prior to entering the Queue.
This patch modifies the behaviour in the following ways:
* If no queue members are paused, the CDR disposition is whatever the
disposition was prior to going into Queue. If the call was answered this
will be ANSWERED; otherwise, it is NO ANSWER.
* If some queue members are pused, the CDR result is NO ANSWER. (This is a
change in behaviour, as the result would previously have been BUSY)
* If all queue members are paused, the CDR result is whatever the result was
prior to going into Queue. This is the same as the behaviour prior to this
patch.
* If the caller hangs up, times out, or presses '*' with the 'h' option, the
CDR disposition is again not set and is dependent on whether or not the
caller was Answered prior to entering Queue.
This patch was based on one provided by Thomas Arimont, but has been modified
to accomodate findings by the reviewers.
Review: https://reviewboard.asterisk.org/r/2064/
(closes issue AST-906)
Reported by: Thomas Arimont
(closes issue ASTERISK-17776)
Reported by: Attila Megyeri
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375416 65c4cc65-6c06-0410-ace0-fbb531ad65f3