AST_LIST_INSERT_BEFORE_CURRENT() could not be used twice in an iteration
or before AST_LIST_REMOVE_CURRENT() without corrupting the list.
AST_LIST_INSERT_BEFORE_CURRENT() could also corrupt the list if
AST_LIST_INSERT_BEFORE_CURRENT() or AST_LIST_REMOVE_CURRENT() is used on
the next iteration.
* Fixed cut and paste error using the wrong variable in
AST_LIST_INSERT_BEFORE_CURRENT().
* Added linked list unit tests for AST_LIST_INSERT_BEFORE_CURRENT(),
AST_LIST_APPEND_LIST(), and AST_LIST_INSERT_LIST_AFTER().
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Sequence number was handled as an unsigned integer (usually 32 bits I think, more
depending on the architecture) and was put into the rtp packet which is basically
just a bunch of bits using an or operation. Sequence number only has 16 bits
allocated to it in an RTP packet anyway, so it would add to the next field which
just happened to be the codec. This makes sure the sequence number is set to be
a 16 bit integer regardless of architecture (hopefully) and also makes it so the
incrementing of the sequence number does bitwise or at the peak of a 16 bit number
so that the value will be set back to 0 when going beyond 65535 anyway.
(closes issue ASTERISK-18291)
Reported by: Will Schick
Review: https://reviewboard.asterisk.org/r/1542/
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The "No D-channels available! Using Primary channel as D-channel anyway!"
WARNING message has been confusing on non-NFAS setups. The message refers
to things that are NFAS specific.
* Changed the warning to several different warnings to be more accurate
for the situation and less confusing as a result:
"No D-channels up! Switching selected D-channel from X to Y.",
"No D-channels up!", and
"D-channel is down!".
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Despite an ominous sounding comment stating that membercount was for "logged
in" members only and thus we couldn't use ao2_container_count(), I could not
find a single place in the code where that seemed to be accurate. The only time
we decremented membercount was when we were marking something dead or actually
removing it. The only places we incremented it were either after ao2_link(), or
trying to correct for having set it to 0 during a reload. In every case where
we were correcting the value, it seemed that we were trying to make the count
actually match what ao2_container_count() would return. The only place I could
find where we made a determination about something being "logged in" or not, we
didn't trust the membercount, but instead looked at devicestate, paused, etc.
This patch removes membercount, replaces its use with ao2_container_count, and
manually adds the results of ao2_container_count to a "membercount" field for
ast_data queue query results. This patch also would fix AST-676, but as it is
slightly riskier than the previously committed fix, the two commits have been
made separately.
Reivew: https://reviewboard.asterisk.org/r/1541/
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r342277 | kmoore | 2011-10-25 11:08:04 -0500 (Tue, 25 Oct 2011) | 25 lines
Merged revisions 342276 via svnmerge from
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r342276 | kmoore | 2011-10-25 11:06:57 -0500 (Tue, 25 Oct 2011) | 18 lines
Fix spool handling to allow call files to be hardlinked into place
This fixes the inotify code to handle call files being hardlinked into the
spool directory.
The smsq utility does this, instead of rename(), to ensure that it cannot
accidentally overwrite an existing spool file. A rename() might do that, but
link() will definitely not.
The inotify code had broken this, because it would wait for an IN_CLOSE_WRITE
event on the file... which was never forthcoming, since it was never opened.
Now we look for IN_OPEN events following the IN_CREATE event, and only wait
for an IN_CLOSE_WRITE if the file was actually opened.
Patch-by: dwmw2
(closes issue ASTERISK-18331)
Review: https://reviewboard.asterisk.org/r/1391/
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To use the new OBJ_KEY flag, the container hash and compare callback
functions must be updated to support OBJ_KEY. Otherwise, bad things
happen.
(issue ASTERISK-14769)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342112 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The fix for ASTERISK-12715 and ASTERISK-12685 added a check for the Park
application because the channel needed to be masqueraded to prevent a
crash. Since the Park application now always masquerades the channel into
the parking lot, the special check is no longer needed. The fix also
resulted in AGI exec Park attempting to double park the call and not honor
the Park application parameters.
* Removed no longer necessary call to ast_masq_park_call() by AGI exec for
the Park application. (Reverts -r146923)
* Fix Park application to only return 0 or -1. The AGI exec Park was
causing broken pipe error messages because the Park application returned 1
on successful park.
(closes issue ASTERISK-18737)
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r341599 | irroot | 2011-10-20 20:20:08 +0200 (Thu, 20 Oct 2011) | 8 lines
add documentation for check_state_unknown in configs/queues.conf.sample
app_queue allows calls to members in a "Unknown" state to be treated as available
setting check_state_unknown = yes will cause app_queue to query the channel driver
to better determine the state this only applies to queues with ringinuse or ignorebusy
set appropriately.
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r341580 | irroot | 2011-10-20 19:13:23 +0200 (Thu, 20 Oct 2011) | 15 lines
Add option to check state when state is unknown
r341486 reverts r325483 this is a rework of the patch.
optimize to minimize load.
add option check_state_unknown to control whether a member with unknown
device state is checked there is a small % chance that calls will be sent
to the member when they on a call.
app_queue will see a device with unknown state as available and does not
try verify the state without this option enabled.
Review: https://reviewboard.asterisk.org/r/1535/
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r341486 | mnicholson | 2011-10-19 16:23:17 -0500 (Wed, 19 Oct 2011) | 18 lines
Fix a performance regression introduced in r325483.
The regression was caused by a call to ast_parse_device_state() in app_queue's
ring_entry() function. The ast_parse_device_state() function eventually calls
ast_channel_get_full() with a channel name prefix which causes it to walk the
channel list causing massive lock contention and slow downs.
This patch fixes the regression by removing the call to
ast_parase_device_state() which should be unnecessary. Queue member device
state should be maintained by device state events. Some users have seen
instances where busy agents were called when they shouldn't have, which is the
reason the call to ast_parse_device_state() was added. That change appears to
have resolved that issue but also causes this performance regression. There may
still be issues with queue member status, and if so, alternative methods should
be investigated to resolve them.
AST-695
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Google has recently make some changes (again) to their protocol. Rather then
patching asterisk to flip between the two different methods, we now allow both.
Lets hope this keeps Google Voice happy for a while.
(closes issue ASTERISK-18714)
Reported by: Iordan Iordanov
Patches:
chan_gtalk.patch uploaded by Iordan Iordanov (licenses 6311)
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* Fix potential deadlocks in SIP and IAX blind transfer to parking.
* Fix SIP, IAX, DAHDI analog, and MGCP channel drivers to respect the
parkext_exclusive option with transfers (Park(,,,,,exclusive_lot)
parameter). Created ast_park_call_exten() and ast_masq_park_call_exten()
to maintian API compatibility.
* Made masq_park_call() handle a failed ast_channel_masquerade() setup.
* Reduced excessive struct parkeduser.peername[] size.
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whether modules are embedded or not; using just the bare category name led to
accidentally enabling these options when users used the wrong "--enable"
operation on the menuselect command line.
Now the internal option names are prefixed with "EMBED_", so they won't be
the same as the name of the category containing the modules they control
the embedding of.
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If a simple switch was started on a device and then a specific call
made (such as redial or speed dial), on timeout of the simple switch
the call would be attempted again. This patch only allows the simple
switch to make a call if the substate is still in the collecting
digits mode.
Also added small debug message to dialAndAactivate sub.
Tested by snuff and myself.
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r340971 | kmoore | 2011-10-14 15:50:37 -0500 (Fri, 14 Oct 2011) | 15 lines
Merged revisions 340970 via svnmerge from
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r340970 | kmoore | 2011-10-14 15:49:39 -0500 (Fri, 14 Oct 2011) | 8 lines
Quiet RTCP Receiver Reports during fax transmission
RTCP is now disabled for "inactive" RTP audio streams during SIP T.38 sessions.
The ability to disable RTCP streams in res_rtp_asterisk was missing, so this
code was added to support the bug fix.
(closes issue ASTERISK-18400)
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Party A calls Party B.
Party A DTMF blind transfers Party B to Party C.
Party A channel continues to execute dialplan.
* Fixed the return value of builtin_blindtransfer() to return the correct
value after a transfer so the dialplan will not keep executing.
* Removed unnecessary connected line update that did not really do
anything.
* Made access to GOTO_ON_BLINDXFR thread safe in check_goto_on_transfer().
* Fixed leak of xferchan for failure cases in check_goto_on_transfer().
* Updated debug messages in builtin_blindtransfer() and
check_goto_on_transfer().
(closes issue ASTERISK-18275)
Reported by: rmudgett
Tested by: rmudgett
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r339463 | irroot | 2011-10-05 08:28:46 +0200 (Wed, 05 Oct 2011) | 9 lines
Only change the capabilities on the gateway when
the session is been destroyed there is still
a race condition that ends in a segfault.
if the caps are changed the logic in res_fax_spandsp
will run T30 code not gateway code to end the session.
this has been experienced on a "slower" under spec system.
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