Re-add support for spaces in pathnames, including now spaces in DESTDIR.
This was initially added to 1.8 prior to release, primarily to support the
standard paths on Mac OS X, but was partially reverted recently in Subversion,
due to the lack of support for spaces in DESTDIR. This commit restores support
for the standard paths on Mac OS X, and also includes support for spaces in
DESTDIR.
Numerous isues have been reported for deadlocks that are caused by
a blocking read in res_timing_timerfd on a file descriptor that will
never be written to. This patch adds some checks to make sure that
the timerfd is both valid and armed before calling read().
Should fix: ASTERISK-18142, ASTERISK-18166, ASTERISK-18197, AST-486,
AST-495, AST-507 and possibly others.
Outgoing BRI calls fail when using Asterisk 1.8 with HA8, HB8, and B410P cards.
France Telecom brings layer 2 and layer 1 down on BRI lines when the line
is idle. When layer 1 goes down Asterisk cannot make outgoing calls and
the HA8 and HB8 cards also get IRQ misses.
The inability to make outgoing calls is because the line is in red alarm
and Asterisk will not make calls over a line it considers unavailable.
The IRQ misses for the HA8 and HB8 card are because the hardware is
switching clock sources from the line which just brought layer 1 down to
internal timing.
There is a DAHDI option for the B410P card to not tell Asterisk that layer
1 went down so Asterisk will allow outgoing calls: "modprobe wcb4xxp
teignored=1". There is a similar DAHDI option for the HA8 and HB8 cards:
"modprobe wctdm24xxp bri_teignored=1". Unfortunately that will not clear
up the IRQ misses when the telco brings layer 1 down.
* Add layer 2 persistence option to customize the layer 2 behavior on BRI
PTMP lines. The new option has three settings: 1) Use libpri default
layer 2 setting. 2) Keep layer 2 up. Bring layer 2 back up when the peer
brings it down. 3) Leave layer 2 down when the peer brings it down.
Layer 2 will be brought up as needed for outgoing calls.
ASTERISK-18067 ASTERISK-15479 - White Space affects mailbox value, multiple MWI subs
Before, having multiple subscriptions to mailboxes on a sip peer set via the mailbox
setting in sip.conf would only result in updates being sent on whichever mailbox
triggered the mwi event. Now all of them get counted regardless. Also fixes a bug
involving parsing of the mailbox option in sip.conf so that trailing and leading
spaces before/after commas are trimmed.
(closes issue ASTERISK-18067)
Reported by: aragon
(closes issue ASTERISK-15479)
Reported by: Ben Winslow
Patches: chan_sip.c-mwi_multi_mailbox_fix-1.6.2.13.diff (License #5288) patch uploaded by Ben Winslow
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-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.
Added the 'storesipcause' option to sip.conf to allow the user to disable the
setting of HASH(SIP_CAUSE,<chan name>) on the channel.
Having chan_sip set HASH(SIP_CAUSE,<chan name>) on the channel carries a
significant performance penalty because of the usage of the MASTER_CHANNEL()
dialplan function.
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
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AMI actions DAHDIHangup and DAHDITransfer have no effect.
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.
Logger does not warn of failure to open logging channels
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.
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.
Output of queue log not started until logger reloaded.
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
Jonathan Rose [Wed, 10 Aug 2011 15:45:57 +0000 (15:45 +0000)]
SIP display-name needed to be empty for Avaya IP500
In order to address a compatability issue with certain features on certain devices
which rely on display name content to change behavior, initreqprep in chan_sip.c
has been changed to no longer substitute cid_number into the display name when
cid_name isn't present. Instead, it will send no display name in that case.
(closes issue ASTERISK-16198)
Reported by: Walter Doekes
AMI action ModuleReload returns Error if Module: missing or empty
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."
* 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.
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Alexandr Anikin [Tue, 9 Aug 2011 17:12:27 +0000 (17:12 +0000)]
Merged revisions 331147,331200 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
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r331147 | may | 2011-08-09 20:16:55 +0400 (Tue, 09 Aug 2011) | 11 lines
Merged revisions 331146 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r331146 | may | 2011-08-09 20:13:09 +0400 (Tue, 09 Aug 2011) | 4 lines
move ast_cond_signal for admitted call after all data filled/freed
clear all log channels by pointed number not only first
free allocated callToken in ooh323_answer
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r331200 | may | 2011-08-09 20:36:39 +0400 (Tue, 09 Aug 2011) | 9 lines
Setup IP proto version for call in GK mode
Added additional check for IP semantics before parse destination
by ast_parse_args due to it can parse numeric as IP.
Kinsey Moore [Tue, 9 Aug 2011 17:08:33 +0000 (17:08 +0000)]
Allow ENUM query functions to report lookup errors
The ENUM dialplan functions do not report DNS query errors properly. It is
useful to differentiate between failed query (e.g. non-existent domain) vs. no
data records of the appropriate type. This is required to make overlapped
dialing work.
As a result of converting Unlink events that were missed in the AMI
1.1 update to Bridge events, the AMI protocol version is being incremented.
........
A previous update converted some of the Link and Unlink events to
Bridge events, but a couple of Unlink events were missed. This patch
rectifies the situation.
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>
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Kinsey Moore [Mon, 8 Aug 2011 20:28:20 +0000 (20:28 +0000)]
Log queue member name when state_interface is set for ADDMEMBER and REMOVEMEMBER events
app_queue logs the events ADDMEMBER and REMOVEMEMBER with the agent field set
to the interface value rather than the membername value when a member is added
with a state_interface value set. However all other member related queue
events are logged with the membername when a state_interface is set. This
patch makes these fields optionally more consistent and correct.
Kinsey Moore [Mon, 8 Aug 2011 15:00:26 +0000 (15:00 +0000)]
app_queue: Add StateInterface to output of "queue show" and "QueueStatus"
This patch adds the state_interface of the queue member struct to the output
of "queue show" (CLI command) and "QueueStatus" (AMI action) when displaying
relevant queue member information. For the AMI event message the variable
StateInterface has been added.
Alexandr Anikin [Fri, 5 Aug 2011 08:47:59 +0000 (08:47 +0000)]
Merged revisions 330899 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
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r330899 | may | 2011-08-05 11:38:28 +0400 (Fri, 05 Aug 2011) | 11 lines
Merged revisions 330827 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r330827 | may | 2011-08-04 23:37:16 +0400 (Thu, 04 Aug 2011) | 4 lines
change gk client behaivour on rrq/grq failures to setup timers
and next tries after timeout instead of complete failure in the ooh323
stack
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editing files in main/editline does not ensure rebuild of libedit.a
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
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Call pickup broken for DAHDI channels when beginning with #
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.
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.
Incorrect playback for Spanish in some circumstances
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
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Russell Bryant [Fri, 29 Jul 2011 19:34:36 +0000 (19:34 +0000)]
astobj2: Avoid using temporary objects + ao2_find() with OBJ_POINTER.
There is a fairly common pattern making its way through the code base where we
put a temporary object on the stack so we can call ao2_find() with OBJ_POINTER.
The purpose is so that it can be passed into the object hash function.
However, this really seems like a hack and potentially error prone. This patch
is a first stab at approach to avoid having to do that.
It adds a new flag, OBJ_KEY, which can be used instead of OBJ_POINTER in these
situations. Then, the hash function can know whether it was given an object or
some custom data to hash.
The patch also changes some uses of ao2_find() for iax2_user and iax2_peer
objects to reflect how OBJ_KEY would be used.
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.
The default conf-usermenu says that '8' can be used to leave the conference, so
put that in the sample user menu. '5' is supposed to extend the conference, but
there doesn't appear to be a concept of that in the menu actions.
........
Changes sound file for prepend "then-press-pound" to "vm-then-pound" which is the same
prompt, only it turned out "then-press-pound" was part of extra sounds. Also, vm is more
appropriate anyway.
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Fixes some voicemail forwarding behavior based around prepend mode.
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.
Remove lastmsgssent from sip it has not been working since 1.6
Clean up the return values to be consistant not currently used
Add doxygen returns
MWI Event is sent on Register
Deadlocks dealing with dialplan hints during reload.
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.
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
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FreeBSD test fails on this case presumably because there is no eth0 on the test
machine. Better to just remove this test for now.
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The functionality of inband DTMF in chan_sip relied upon
ast_rtp_instance_dtmf_mode_get/set not working properly to avoid calling
ast_rtp_instance_dtmf_begin/end on RTP streams with inband DTMF. According to
documentation, ast_rtp_instance_dtmf_begin/end is meant only for RFC2833 DTMF,
never inband. This fixes the regression introduced in revision 328823.
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Revert partial attempt at handling pathnames with spaces.
Revision 299794 attempted to improve the build system to be able to handle
pathnames (primarily DESTDIR) with spaces in them, since this is common on
some platforms (including Mac OSX). Unfortunately, the changes were incomplete
and did not actually provide the desired behavior, and as a side effect the
functionality that ensured that stale headers in the Asterisk 'include' directory
were removed got broken. In addition, the check for stale (and possibly
incompatible) modules in the Asterisk 'modules' directory also got broken, and
would never report any stale modules. Users upgrading to this version or later
versions would then see unexpected module load errors.
Since there are few users who actually want to install Asterisk into paths
that contain spaces, and a proper fix for the build system would take many hours,
the best solution for now is to just revert the partial solution.
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RTP bridge away with inband DTMF and feature detection
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.
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.
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
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projects / thirdparty / asterisk.git / log
