Git does not support the ability to replace a token with a version
string during check-in. While it does have support for replacing a
token on clone, this is somewhat sub-optimal: the token is replaced
with the object hash, which is not particularly easy for human
consumption. What's more, in practice, the source file version was often
not terribly useful. Generally, when triaging bugs, the overall version
of Asterisk is far more useful than an individual SVN version of a file. As a
result, this patch removes Asterisk's support for showing source file
versions.
Specifically, it does the following:
* Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and
remove passing the version in with the macro. Other facilities
than 'core show file version' make use of the file names, such as
setting a debug level only on a specific file. As such, the act of
registering source files with the Asterisk core still has use. The
macro rename now reflects the new macro purpose.
* main/asterisk:
- Refactor the file_version structure to reflect that it no longer
tracks a version field.
- Remove the "core show file version" CLI command. Without the file
version, it is no longer useful.
- Remove the ast_file_version_find function. The file version is no
longer tracked.
- Rename ast_register_file_version/ast_unregister_file_version to
ast_register_file/ast_unregister_file, respectively.
* main/manager: Remove value from the Version key of the ModuleCheck
Action. The actual key itself has not been removed, as doing so would
absolutely constitute a backwards incompatible change. However, since
the file version is no longer tracked, there is no need to attempt to
include it in the Version key.
* UPGRADE: Add notes for:
- Modification to the ModuleCheck AMI Action
- Removal of the "core show file version" CLI command
Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
When an IMAP backend is in use and greetings are set to be used, but aren't
present for a user in their IMAP folder, Asterisk will crash. This occurs
due to the mailstream being set to the 'greetings' folder and being left
in that particular state, regardless of the success/failure of the attempt
to access the folder the mailstream points to. Later access of the mailstream
assumes that it points to the 'INBOX' (or some other folder), resulting in
either a crash (if the greetings folder didn't exist and the mailstream is
invalid) or an inability to read messages from the 'INBOX' folder.
This patch restores the mailstream to its correct state after accessing the
greetings. This fixes the crash, and sets the mailstream to the state that
VoiceMailMain expects.
Note that while ASTERISK-23390 also contained a patch for this issue, the
patch on ASTERISK-24786 is the one being merged here.
Review: https://reviewboard.asterisk.org/r/4459/
ASTERISK-23390 #close
Reported by: Ben Smithurst
ASTERISK-24786 #close
Reported by: Graham Barnett
Tested by: Graham Barnett
patches:
app_voicemail.c.patch.SIGSEGV3rev2 uploaded by Graham Barnett (License 6685)
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When using IMAP voicemail with FreePBX, you will often get ERROR messages
complaining about not being able to find a mailbox. This is due to how FreePBX
handles voicemail mailboxes. Unfortunately, app_voicemail has to consider this
a configuration error, as in any other system it would be indicative of
someone misconfiguring their system.
Regardless, a misconfiguration is a WARNING, and not an ERROR. This patch
demotes the message so that system administrators can hopefully reduce some
of the noise in their log files.
Note that in the original patch this was made into a NOTICE, but that's a
too forgiving.
ASTERISK-24790 #close
Reported by: Graham Barnett
patches:
app_voicemail.c.patch_noise uploaded by Graham Barnett (License 6685)
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When interfacing with Microsoft Exchange, custom headers will be returned as
all lower case. Currently, the IMAP header code will fail to parse the returned
custom headers, as it will be performing a case sensitive comparison. This can
cause playback of messages to fail, as needed information - such as origtime -
will not be present.
This patch updates app_voicemail's header parsing code to perform a case
insensitive lookup for the requested custom headers. Since the headers are
specific to Asterisk, e.g., 'x-asterisk-vm-orig-time', and headers should be
unique in an IMAP message, this should cause no issues with other systems.
ASTERISK-24787 #close
Reported by: Graham Barnett
patches:
app_voicemail.c.patch_MSExchange uploaded by Graham Barnett (License 6685)
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The MixMonitor m() option allows a recording to be pushed to a specific
voicemail mailbox. If the message is delivered to the mailbox's INBOX, however,
no MWI notification is currently raised.
This patch corrects the issue by properly calling notify_new_state from the
msg_create_from_file function. This will cause MWI to be triggered if the
message was placed in the mailbox's INBOX.
ASTERISK-24709 #close
Reported by: Gareth Palmer
patches:
app_voicemail-430919.patch uploaded by Gareth Palmer (License 5169)
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When using ODBC or IMAP storage, temporary files created on the file system
must be disposed of using the DISPOSE macro. The DELETE macro will map to a
deletion function for the backend storage, but does not clean up any local
files created as a result of the operation.
When using voicemail with the operator and review options enabled, pressing
0 to enter the menu, followed by 1 to save the message, followed by any
other DTMF press to delete the message, will result in the temporary file
lingering on the file system.
This patch properly calls DISPOSE after the DELETE. This causes the local
file to be disposed of.
ASTERISK-24288 #close
Reported by: LEI FU
patches:
voicemail_odbc_review_fix.diff uploaded by LEI FU (License 6640)
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* Made the following AMI actions use list API calls for consistency:
Agents
BridgeInfo
BridgeList
BridgeTechnologyList
ConfbridgeLIst
ConfbridgeLIstRooms
CoreShowChannels
DAHDIShowChannels
DBGet
DeviceStateList
ExtensionStateList
FAXSessions
Hangup
IAXpeerlist
IAXpeers
IAXregistry
MeetmeList
MeetmeListRooms
MWIGet
ParkedCalls
Parkinglots
PJSIPShowEndpoint
PJSIPShowEndpoints
PJSIPShowRegistrationsInbound
PJSIPShowRegistrationsOutbound
PJSIPShowResourceLists
PJSIPShowSubscriptionsInbound
PJSIPShowSubscriptionsOutbound
PresenceStateList
PRIShowSpans
QueueStatus
QueueSummary
ShowDialPlan
SIPpeers
SIPpeerstatus
SIPshowregistry
SKINNYdevices
SKINNYlines
Status
VoicemailUsersList
* Incremented the AMI version to 2.7.0.
* Changed astman_send_listack() to not use the listflag parameter and
always set the value to "Start" so the start capitalization is consistent.
i.e., The FAXSessions used "Start" while the rest of the system used
"start". The corresponding complete event always used "Complete".
* Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the
AMI ActionID for all of its list events.
* Fixed off-nominal AMI protocol error in manager_bridge_info(),
manager_parking_status_single_lot(), and
manager_parking_status_all_lots(). Use of astman_send_error() after
responding to the original AMI action request violates the action response
pattern by sending two responses.
* Fixed minor protocol error in action_getconfig() when no requested
categories are found. Each line needs to be formatted as "Header: text".
* Fixed off-nominal memory leak in manager_build_parked_call_string().
* Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail().
ASTERISK-24049 #close
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/4315/
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The UW IMAP library is instrinsically not thread-safe, and relies upon higher
level applications to guarantee thread safety. For the most part, this is
provided by the vms object, which provides locking for individual streams.
Unfortunately, this is not sufficient for calls to mail_open which create the
IMAP stream. mail_open can, on some systems, call into a UW IMAP specific
function for determining the address of a system based on a hostname,
ip_nametoaddr.
In the ip6_unix implementation of this function, static variables are used
to hold parsing buffers. This can cause a crash if multiple threads attempt
to convert a hostname to an address at the same time. Locking on a single
mail stream is not sufficient to prevent simultaneous access to these static
variables.
In the IMAP library, this function can be called from the mail_open and
imap_status functions. As the imap_status function is not used by
app_voicemail, locking on access to mail_open is sufficient to prevent
any mangling of the buffers.
Review: https://reviewboard.asterisk.org/r/4188/
ASTERISK-24516 #close
Reported by: David Duncan Ross Palmer
Tested by: David Duncan Ross Palmer
patches:
ASTERISK-24516.diff uploaded by David Duncan Ross Palmer (License 6660)
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An issue existed in r420577, which added multiple recipients to voicemail
emails. The patch, when looking at the intended recipients, looked ahead for
the '|' character inside a while loop which already had pulled out the
appropriate field parsing on the '|' character. This would cause it to skip
the recipients.
This patch fixes it such that it relies completely on the while loop to parse
through the e-mail fields.
Note that the original author of the patch looked at this fix and approved it.
ASTERISK-24250 #close
Reported by: abelbeck
patches:
voicemail-420577-to-comma-fix.diff uploaded by abelbeck (License 5903)
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This patch provides the capability to manipulate templates and categories
with non-unique names via AMI.
Summary of changes:
GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list
of name_regex=value_regex expressions which will cause only categories whose
variables match all expressions to be considered. The special variable name
TEMPLATES can be used to control whether templates are included. Passing
'include' as the value will include templates along with normal categories.
Passing 'restrict' as the value will restrict the operation to ONLY templates.
Not specifying a TEMPLATES expression results in the current default behavior
which is to not include templates.
UpdateConfig: NewCat now includes options for allowing duplicate category
names, indicating if the category should be created as a template, and
specifying templates the category should inherit from. The rest of the
actions now accept a filter string as defined above. If there are non-unique
category names, you can now update specific ones based on variable values.
To facilitate the new capabilities in manager, corresponding changes had to be
made to config, most notably the addition of filter criteria to many of the
APIs. In some cases it was easy to change the references to use the new
prototype but others would have required touching too many files for this
patch so a wrapper with the original prototype was created. Macros couldn't
be used in this case because it would break binary compatibility with modules
such as res_digium_phone that are linked to real symbols.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4033/
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Changes made during format improvements resulted in the
recording to voicemail option 'm' of the MixMonitor app
writing a zero length duration in the msgXXXX.txt file.
This change introduces a new function ast_ratestream(),
which provides the sample rate of the format associated
with the stream, and updates the app_voicemail function
for ast_app_copy_recording_to_vm to calculate the right
duration.
Review: https://reviewboard.asterisk.org/r/3996/
ASTERISK-24328 #close
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This patch adds support for the Japanese language to both the say family of
applications, as well as for VoiceMail and VoiceMailMain. A new pack of
language sounds will be released at the same time as the next major version
of Asterisk to support the new language features.
The language features can be enabled using a language code of 'ja'.
Review: https://reviewboard.asterisk.org/r/3477
ASTERISK-23324 #close
Reported by: Kevin McCoy
patches:
app_voicemail.c.20140226.jb.patch uploaded by Kevin McCoy (License 6586)
say.c.20140226.jb.patch uploaded by Kevin McCoy (License 6586)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417591 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Extract the sayname API call to its own registerd callback. This allows
the app_directory and app_chanspy applications to say a mailbox owner's
name using an alternate provider when app_voicemail is not available
because you are using res_mwi_external. app_directory still uses the
voicemail.conf file.
AFS-64 #close
Reported by: Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch addresses some aesthetic issues in Asterisk. These are all just
minor tweaks to improve the look of the CLI when used in a variety of
settings. Specifically:
* A number of chatty verbose messages were removed or demoted to DEBUG
messages. Verbose messages with a verbosity level of 5 or higher were -
if kept as verbose messages - demoted to level 4. Several messages
that were emitted at verbose level 3 were demoted to 4, as announcement
of dialplan applications being executed occur at level 3 (and so the
effects of those applications should generally be less).
* Some verbose messages that only appear when their respective 'debug'
options are enabled were bumped up to always be displayed.
* Prefix/timestamping of verbose messages were moved to the verboser
handlers. This was done to prevent duplication of prefixes when the
timestamp option (-T) is used with the CLI.
* Verbose magic is removed from messages before being emitted to
non-verboser handlers. This prevents the magic in multi-line verbose
messages (such as SIP debug traces or the output of DumpChan) from
being written to files.
* _Slightly_ better support for the "light background" option (-W) was
added. This includes using ast_term_quit in the output of XML
documentation help, as well as changing the "Asterisk Ready" prompt to
bright green on the default background (which stands a better chance of
being displayed properly than bright white).
Review: https://reviewboard.asterisk.org/r/3547/
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Much needed was a way to assign id to objects on creation, and
much change was necessary to accomplish it. Channel uniqueids
and linkedids are split into separate string and creation time
components without breaking linkedid propgation. This allowed
the uniqueid to be specified by the user interface - and those
values are now carried through to channel creation, adding the
assignedids value to every function in the chain including the
channel drivers. For local channels, the second channel can be
specified or left to default to a ;2 suffix of first. In ARI,
bridge, playback, and snoop objects can also be created with a
specified uniqueid.
Along the way, the args order to allocating channels was fixed
in chan_mgcp and chan_gtalk, and linkedid is no longer lost as
masquerade occurs.
(closes issue ASTERISK-23120)
Review: https://reviewboard.asterisk.org/r/3191/
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* The core external MWI resource provides for MWI message counts
persistence using sorcery. With sorcery, the user is able to configure
which sorcery wizzard backend to use if the default astdb is not desired.
* The core external MWI resoruce provides some debugging CLI commands
enabled by defining MWI_DEBUG_CLI.
The debugging CLI commands are:
"mwi delete all",
"mwi delete like <regex>",
"mwi delete mailbox <mailbox>",
"mwi list all",
"mwi list like <regex>",
"mwi show mailbox <mailbox>", and
"mwi update mailbox <mailbox> [<new> [<old>]]".
(closes issue AFS-43)
Review: https://reviewboard.asterisk.org/r/3061/
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This change is in preparation for external MWI support.
Removed code from the system for normal mailbox handling that appends
@default to the mailbox identifier if it does not have a context. The
only exception is the legacy hasvoicemail users.conf option. The legacy
option will only work for app_voicemail mailboxes. The system cannot make
any assumptions about the format of the mailbox identifer used by
app_voicemail.
chan_sip and chan_dahdi/sig_pri had the most changes because they both
tried to interpret the mailbox identifier. chan_sip just stored and
compared the two components. chan_dahdi actually used the box
information.
The ISDN MWI support configuration options had to be reworked because
chan_dahdi was parsing the box@context format to get the box number. As a
result the mwi_vm_boxes chan_dahdi.conf option was added and is documented
in the chan_dahdi.conf.sample file.
Review: https://reviewboard.asterisk.org/r/3072/
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Original commit message by mmichelson (asterisk 12 r403311):
"This adds channel locks around calls to create channel snapshots as well
as other functions which operate on a channel and then end up
creating a channel snapshot. Functions that expect the channel to be
locked prior to being called have had their documentation updated to
indicate such."
The above was initially committed and then reverted at r403398. The problem
was found to be in core_local.c in the publish_local_bridge_message function.
The ast_unreal_lock_all function locks and adds a reference to the returned
channels and while they were being unlocked they were not being unreffed when
no longer needed. Fixed by unreffing the channels.
Also in bridge.c a lock was obtained on "other->chan", but then an attempt was
made to unlock "other" and not the previously locked channel. Fixed by
unlocking "other->chan"
(closes issue ASTERISK-22709)
Reported by: John Bigelow
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* The voicemail registration/unregistration functions now take a struct of
callbacks instead of a lengthy parameter list of callbacks.
* The voicemail registration/unregistration functions now prevent a
competing module from interfering with an already registered callback
supplying module.
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This adds channel locks around calls to create channel snapshots as well
as other functions which operate on a channel and then end up
creating a channel snapshot. Functions that expect the channel to be
locked prior to being called have had their documentation updated to
indicate such.
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In the last release of sounds, 1.4.25 we added a vm-msgforwarded prompt for various core languages. Now we use that prompt.
(issue ASTERISK-21413)
(closes issue ASTERISK-21413)
Reported by: netwrkr
Tested by: newtonr
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r399887 | dlee | 2013-09-26 10:41:47 -0500 (Thu, 26 Sep 2013) | 1 line
Minor performance bump by not allocate manager variable struct if we don't need it
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r400138 | dlee | 2013-09-30 10:24:00 -0500 (Mon, 30 Sep 2013) | 23 lines
Stasis performance improvements
This patch addresses several performance problems that were found in
the initial performance testing of Asterisk 12.
The Stasis dispatch object was allocated as an AO2 object, even though
it has a very confined lifecycle. This was replaced with a straight
ast_malloc().
The Stasis message router was spending an inordinate amount of time
searching hash tables. In this case, most of our routers had 6 or
fewer routes in them to begin with. This was replaced with an array
that's searched linearly for the route.
We more heavily rely on AO2 objects in Asterisk 12, and the memset()
in ao2_ref() actually became noticeable on the profile. This was
#ifdef'ed to only run when AO2_DEBUG was enabled.
After being misled by an erroneous comment in taskprocessor.c during
profiling, the wrong comment was removed.
Review: https://reviewboard.asterisk.org/r/2873/
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r400178 | dlee | 2013-09-30 13:26:27 -0500 (Mon, 30 Sep 2013) | 24 lines
Taskprocessor optimization; switch Stasis to use taskprocessors
This patch optimizes taskprocessor to use a semaphore for signaling,
which the OS can do a better job at managing contention and waiting
that we can with a mutex and condition.
The taskprocessor execution was also slightly optimized to reduce the
number of locks taken.
The only observable difference in the taskprocessor implementation is
that when the final reference to the taskprocessor goes away, it will
execute all tasks to completion instead of discarding the unexecuted
tasks.
For systems where unnamed semaphores are not supported, a really
simple semaphore implementation is provided. (Which gives identical
performance as the original taskprocessor implementation).
The way we ended up implementing Stasis caused the threadpool to be a
burden instead of a boost to performance. This was switched to just
use taskprocessors directly for subscriptions.
Review: https://reviewboard.asterisk.org/r/2881/
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r400180 | dlee | 2013-09-30 13:39:34 -0500 (Mon, 30 Sep 2013) | 28 lines
Optimize how Stasis forwards are dispatched
This patch optimizes how forwards are dispatched in Stasis.
Originally, forwards were dispatched as subscriptions that are invoked
on the publishing thread. This did not account for the vast number of
forwards we would end up having in the system, and the amount of work it
would take to walk though the forward subscriptions.
This patch modifies Stasis so that rather than walking the tree of
forwards on every dispatch, when forwards and subscriptions are changed,
the subscriber list for every topic in the tree is changed.
This has a couple of benefits. First, this reduces the workload of
dispatching messages. It also reduces contention when dispatching to
different topics that happen to forward to the same aggregation topic
(as happens with all of the channel, bridge and endpoint topics).
Since forwards are no longer subscriptions, the bulk of this patch is
simply changing stasis_subscription objects to stasis_forward objects
(which, admittedly, I should have done in the first place.)
Since this required me to yet again put in a growing array, I finally
abstracted that out into a set of ast_vector macros in
asterisk/vector.h.
Review: https://reviewboard.asterisk.org/r/2883/
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r400181 | dlee | 2013-09-30 13:48:57 -0500 (Mon, 30 Sep 2013) | 28 lines
Remove dispatch object allocation from Stasis publishing
While looking for areas for performance improvement, I realized that an
unused feature in Stasis was negatively impacting performance.
When a message is sent to a subscriber, a dispatch object is allocated
for the dispatch, containing the topic the message was published to, the
subscriber the message is being sent to, and the message itself.
The topic is actually unused by any subscriber in Asterisk today. And
the subscriber is associated with the taskprocessor the message is being
dispatched to.
First, this patch removes the unused topic parameter from Stasis
subscription callbacks.
Second, this patch introduces the concept of taskprocessor local data,
data that may be set on a taskprocessor and provided along with the data
pointer when a task is pushed using the ast_taskprocessor_push_local()
call. This allows the task to have both data specific to that
taskprocessor, in addition to data specific to that invocation.
With those two changes, the dispatch object can be removed completely,
and the message is simply refcounted and sent directly to the
taskprocessor.
Review: https://reviewboard.asterisk.org/r/2884/
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Merged revisions 399887,400138,400178,400180-400181 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400186 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds a new dialplan application, SayAlphaCase, that performs much
the same function as SayAlpha except that it takes additional options
which allow the user to specify whether the case of each letter should
be announced for uppercase, lowercase, or all letters. Similar
functionality has been added to the SAY ALPHA AGI command via an
optional parameter.
Original Patch by: Kevin Scott Adams
Reported by: Kevin Scott Adams
Review: https://reviewboard.asterisk.org/r/2725/
(closes issue ASTERISK-20782)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This removes unused code, event types, IE pltypes, and event IE types
where possible and makes several functions private that were once
public. This includes a renumbering of the remaining event and IE types
which breaks binary compatibility with previous versions. The last
remaining consumers of the old event system (or parts thereof) are
main/security_events.c, res/res_security_log.c, tests/test_cel.c,
tests/test_event.c, main/cel.c, and the CEL backends.
Review: https://reviewboard.asterisk.org/r/2703/
(closes issue ASTERISK-22139)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch implements the controls from ARI recordings. The controls
are:
* DELETE /recordings/live/{recordingName} - stop recording and
discard it
* POST /recordings/live/{recordingName}/stop - stop recording
* POST /recordings/live/{recordingName}/pause - pause recording
* POST /recordings/live/{recordingName}/unpause - resume recording
* POST /recordings/live/{recordingName}/mute - mute recording (record
silence to the file)
* POST /recordings/live/{recordingName}/unmute - unmute recording.
Since this underlying functionality did not already exist, is was
added to app.c by a set of control frames, similar to how playback
control works. The pause/mute control frames are toggles, even though
the ARI controls are idempotent, to be consistent with the playback
control frames.
(closes issue ASTERISK-22181)
Review: https://reviewboard.asterisk.org/r/2697/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In working with res_stasis, I discovered a significant limitation to
the current structure of stasis_caching_topics: you cannot subscribe
to cache updates for a single channel/bridge/endpoint/etc.
To address this, this patch splits the cache away from the
stasis_caching_topic, making it a first class object. The stasis_cache
object is shared amongst individual stasis_caching_topics that are
created per channel/endpoint/etc. These are still forwarded to global
whatever_all_cached topics, so their use from most of the code does
not change.
In making these changes, I noticed that we frequently used a similar
pattern for bridges, endpoints and channels:
single_topic ----------------> all_topic
^
|
single_topic_cached ----+----> all_topic_cached
|
+----> cache
This pattern was extracted as the 'Stasis Caching Pattern', defined in
stasis_caching_pattern.h. This avoids a lot of duplicate code between
the different domain objects.
Since the cache is now disassociated from its upstream caching topics,
this also necessitated a change to how the 'guaranteed' flag worked
for retrieving from a cache. The code for handling the caching
guarantee was extracted into a 'stasis_topic_wait' function, which
works for any stasis_topic.
(closes issue ASTERISK-22002)
Review: https://reviewboard.asterisk.org/r/2672/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395954 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Matt Jordan
George Joseph
Walter Doekes
Richard Mudgett
Corey Farrell
Scott Griepentrog
Joshua Colp
Jason Parker
Mark Michelson
Kinsey Moore
Kevin Harwell
David M. Lee
Rusty Newton
Jonathan Rose