Age | Commit message (Collapse) | Author |
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objects
Under some conditions, the memory allocation for stringfields and ao2 objects would not have supplied valid file/function names for MALLOC_DEBUG tracking, so this commit corrects that.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192357 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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As in previous commits, when channels are allocated (with ast_channel_alloc) or datastores are allocated (with ast_datastore_alloc) properly account for the memory being owned by the caller, instead of the allocator function itself.
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accounted in MALLOC_DEBUG mode
This commit modifies the stringfield pool allocator to remember the 'owner' of the stringfield manager the pool is being allocated for, and ensures that pools allocated in the future when fields are populated are owned by that file/function.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192279 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #14985)
Reported by: nikkk
Patches:
20090428__bug14985.diff.txt uploaded by tilghman (license 14)
20090429__bug14985__1.6.1.diff.txt uploaded by tilghman (license 14)
Tested by: nikkk
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192171 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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MALLOC_DEBUG is used
This commit ensures that all astobj2 allocated objects are properly accounted for in MALLOC_DEBUG mode by passing down the file/function/line information from the module/function that actually called the astobj2 allocation function.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192059 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch add a new payload type for information elements, a set
of bit flags. The payload is transported as a 32-bit unsigned integer
but when matching is performed between events and subscribers,
the matching is done by using a bitwise AND instead of numeric value
comparison.
Review: http://reviewboard.asterisk.org/r/242/
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In discussions today at the Europe Asterisk Developer Meet-Up, we determined that
the event_log was used in only 9 places in the entire tree, and really was not needed
at all. The users have been converted to use LOG_NOTICE, or the messages have been
removed since other messages were already in place that provided the same information.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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DAHDI channels
Adds ability for CHANNEL() dialplan function, when used on DAHDI channels,
to temporarily change the number of buffers and/or the buffer policy, and also
to enable, disable, or switch the echo canceller between FAX/data and voice
modes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Reported by Andrew Lindh via the -dev list.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191367 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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chan_sip allows for outbound TLS connections, but does not allow the user to specify what protocol to use (default was SSLv2, and still is if this new option is not specified). This patch lets the user pick the SSL/TLS client method for outbound connections in sip.
(closes issue #14770)
Reported by: TheOldSaint
(closes issue #14768)
Reported by: TheOldSaint
Review: http://reviewboard.digium.com/r/240/
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This branch adds additional methods to dialplan functions, whereby the result
buffers are now dynamic buffers, which can be expanded to the size of any
result. No longer are variable substitutions limited to 4095 bytes of data.
In addition, the common case of needing buffers much smaller than that will
enable substitution to only take up the amount of memory actually needed.
The existing variable substitution routines are still available, but users
of those API calls should transition to using the dynamic-buffer APIs.
Reviewboard: http://reviewboard.digium.com/r/174/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191140 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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ast_tls_read_conf() is a new api call for handling SSL/TLS options across all conf files. Before this change, SSL/TLS options were not consistent. http.conf and manager.conf required the 'ssl' prefix while sip.conf used options with the 'tls' prefix. While the options had different names in different conf files, they all did the exact same thing. Now, instead of mixing 'ssl' or 'tls' prefixes to do the same thing depending on what conf file you're in, all SSL/TLS options use the 'tls' prefix. For example. 'sslenable' in http.conf and manager.conf is now 'tlsenable' which matches what already existed in sip.conf. Since this has the potential to break backwards compatibility, previous options containing the 'ssl' prefix still work, but they are no longer documented in the sample.conf files. The change is noted in the CHANGES file though.
Review: http://reviewboard.digium.com/r/237/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(issue #14981)
Reported by: snuffy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190989 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r190721 | kpfleming | 2009-04-27 14:29:46 -0500 (Mon, 27 Apr 2009) | 7 lines
Fix 'inconsistent line endings' when autoconf 2.63 is used
Attempt to make configure script regeneration 'safe' using autoconf 2.63, which embeds a bare CR into the script, thus making Subversion complain about inconsistent line endings
This commit changes the MIME type of the configure script to be 'binary' thus making Subversion no longer inspect line endings, and as a bonus 'svn diff' will no longer try to generate diff output for it, which is not generally useful anyway.
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Adds option to specify a private key .pem file when configuring TLS or SSL in AMI, HTTP, and SIP. Before this, the certificate file was used for both the public and private key. It is possible for this file to hold both, but most configurations allow for a separate private key file to be specified. Clarified in .conf files how these options are to be used. The current conf files do not explain how the private key is handled at all, so without knowledge of Asterisk's TLS implementation, it would be hard to know for sure what was going on or how to set it up.
Review: http://reviewboard.digium.com/r/234/
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There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #10961)
Reported by: ys
Patches:
digest_auth_r148468_v5.diff uploaded by ys (license 281)
SVN branch http://svn.digium.com/svn/asterisk/team/group/manager_http_auth
Tested by: ys, twilson, tilghman
Review: http://reviewboard.digium.com/r/223/
Reviewed by: tilghman,russellb,mmichelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r190092 | tilghman | 2009-04-22 16:35:03 -0500 (Wed, 22 Apr 2009) | 7 lines
Detect availability of pthread_rwlock_timedwrlock() before using it.
(closes issue #14930)
Reported by: tilghman
Patches:
20090420__bug14930.diff.txt uploaded by tilghman (license 14)
Tested by: mvanbaak, tilghman
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There seems to be a bug with old versions of g++ that doesn't allow a structure
member to use the name list. Rename list member to group_list in ast_group_info
and change the few places it is used.
(closes issue #14790)
Reported by: stuarth
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r189601 | dbailey | 2009-04-21 09:00:55 -0500 (Tue, 21 Apr 2009) | 3 lines
Add check in configure script to check for GLOB_NOMAGIC and GLOB_BRACE in glob.h
This allows config.c to compile when linked against uclibc that does not support these parameters
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This is the companion commit to libpri r732. Service messages are now supported
for switch types 4ess/5ess. A new option service_message_support has been added
to chan_dahdi.conf and is noted in the sample config file. The service message
support is turned off by default. The current implementation relies on AstDB
to keep track of channel state, which allows the statuses to be preserved
across Asterisk restarts. Below is a description of the storage format.
The state and reason for the service state are in the form <state>:<reason>,
where:
<state> ::= { 'O' } // 'O' – Out Of Service
<reason> ::= { '0' | '1' | '2' | '3' }, where:
'0' – No reason (backwards compatibility)
'1' – NEAR END
'2' – FAR END
'3' – both NEAR and FAR END
The new CLI commands to handle channel service state are:
pri service disable channel <chan>
pri service enable channel <chan>
Many people contributed to the development of this functionality. Because I
entered at the very end I do not know the exact history. Special thanks to
all who moved the bug forward one way or another:
cmaj, PCadach, markster, mattf, drmac, MikeJ, serge-v, murf, kanelbullar, Seb7,
tilghman, lmadsen, and especially dhubbard (he answered lots of my questions
and did a large portion of the work)
(closes issue #3450)
Reported by: cmaj
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much better solution will be used
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The F option to app_dial has been modified to accept no parameters and perform
the above functionality. I don't see anywhere else that is doing function
overloading, but this really is the best place for this operation because:
- It makes it close to the 'g' option in the argument list which provides
similar functionality.
- The existing code to support the current F option provides a very
convienient location to add this new feature.
(closes issue #12381)
Reported by: michael-fig
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187491 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r187428 | tilghman | 2009-04-09 13:08:20 -0500 (Thu, 09 Apr 2009) | 8 lines
Race condition between ast_cli_command() and 'module unload' could cause a deadlock.
Add lock timeouts to avoid this potential deadlock.
(closes issue #14705)
Reported by: jamessan
Patches:
20090320__bug14705.diff.txt uploaded by tilghman (license 14)
Tested by: jamessan
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This was accomplished using a set of options and the setoption channel callback.
The core calls into the channel driver using these options and the channel driver
either returns success or failure.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187360 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r187300 | tilghman | 2009-04-08 23:31:38 -0500 (Wed, 08 Apr 2009) | 3 lines
Add debugging mode for diagnosing file descriptor leaks.
(Related to issue #14625)
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r187301 | tilghman | 2009-04-08 23:32:40 -0500 (Wed, 08 Apr 2009) | 2 lines
Oops, missed this file in the last commit.
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events, so that they can be fed into external processes for analysis and possible mitigation efforts
(inspired by this evening's Toronto Asterisk Users Group meeting and previous dicussions amongst various community members)
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The biggest change done here was elimination of the backup_config for use with
features. Previously, the bridging code upon detecting a feature would set the
start time of the bridge to the start time of the feature. Then after the
feature had either expired or timed out the start time would be reset to the
true bridge start time from the backup_config. Now, the time differences are
calculated with respect to the newly added feature_start_time timeval instead.
There should be no behavior changes from the previous functionality aside from
the bridge timing being unaffected by either valid or partial feature matches.
Previously the timing would be increased by the length of time configured for
featuredigittimeout, which was probably never noticed.
(closes issue #14503)
Reported by: KNK
Tested by: jpeeler
Review: http://reviewboard.digium.com/r/179/
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doxyref.h was created to hold miscellaneous documentation that was not specific
to a part of the code. This file has grown quite a bit so I decided to start
splitting parts of it out into new files. Now, you can drop a new file into
include/asterisk/doxygen/ and it will be processed by doxygen.
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Asterisk.
The channel drivers which have been most heavily tested with these enhancements are
chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be
introduced in a later commit. chan_skinny has code added to it here, but according
to user pj, the support on chan_skinny is not working as of now. This will be fixed in
a later commit.
A special thanks goes out to bugtracker user gareth for getting the ball rolling and
providing the initial support for this work. Without his initial work on this, this would
not have been nearly as painless as it was.
This functionality has been tested by Digium's product quality department, as well as a
customer site running thousands of calls every day. In addition, many many many many bugtracker
users have tested this, too.
(closes issue #8824)
Reported by: gareth
Review: http://reviewboard.digium.com/r/201
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call.
This API call now waits for a special frame from the underlying channel driver to
indicate success or failure. This allows the return value to truly convey whether
the transfer worked or not. In the case of the Transfer() dialplan application this
means the value of the TRANSFERSTATUS dialplan variable is actually true.
(closes issue #12713)
Reported by: davidw
Tested by: file
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r186320 | file | 2009-04-03 12:48:56 -0300 (Fri, 03 Apr 2009) | 5 lines
Fix a problem with the crypto variable definitions not actually being defined properly.
(closes issue #14804)
Reported by: jvandal
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This API provides a generic way for multiple RTP stacks to be
integrated into Asterisk. Right now there is only one present, res_rtp_asterisk,
which is the existing Asterisk RTP stack. Functionality wise this commit
performs the same as previously. API documentation can be viewed in the
rtp_engine.h header file.
Review: http://reviewboard.digium.com/r/209/
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Included is a small bugfix to an ast_str helper, but most of these changes
are simply doxygen fixes.
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This patch provides a number of optimizations to the stringfields API, focused around saving (not wasting) memory whenever possible. Thanks to Mark Michelson for inspiring this work and coming up with the first two optimizations that are represented here:
Changes:
- Cleanup of some code, fix incorrect doxygen comments
- When a field is emptied or replaced with a new allocation, decrease the amount of 'active' space in the pool it was held in; if that pool reaches zero active space, and is not the current pool, then free it as it is no longer in use
- When allocating a pool, try to allocate a size that will fit in a 'standard' malloc() allocation without wasting space
- When allocating space for a field, store the amount of space in the two bytes immediately preceding the field; this eliminates the need to call strlen() on the field when overwriting it, and more importantly it 'remembers' the amount of space the field has available, even if a shorter string has been stored in it since it was allocated
- Don't automatically double the size of each successive pool allocated; it's wasteful
http://reviewboard.digium.com/r/165/
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The doxygen documentation has now been updated to state explicitly that I want
punctuation atthe end of the first sentence in a commit message. :).
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The ability to load/unload timing interfaces is nice, but it means that when a timer is allocated, it may come from provider A, but later provider B becomes the 'preferred' provider. If this happens, all timer API calls on the timer that was provided by provider A will actually be handed to provider B, which will say WTF and return an error.
This patch changes the timer API to include a pointer to the provider of the timer handle so that future operations on the timer will be forwarded to the proper provider.
(closes issue #14697)
Reported by: moy
Review: http://reviewboard.digium.com/r/211/
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When dvossel and I were doing some load testing last week, we noticed that we
could make Asterisk trunk lock up instantly when we started generating a bunch
of calls. The backtraces of locked threads were bizarre, and many were stuck
on an _unlock_ of an rwlock.
The changes are:
1) Fix a number of places where a backtrace would be loaded into an invalid
index of the backtrace array. It's an off by one error, which ends up
writing over the rwlock itself.
2) Ensure that in the array of held locks, we NULL out an index once it is
not being used so that it's not confusing when analyzing its contents.
3) Remove a bunch of logging referring to an rwlock operating being done
with "deep reentrancy". It is normal for _many_ threads to hold a
read lock on an rwlock.
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This code comes from svn/asterisk/team/russell/event_performance/.
Here is a summary of the changes that have been made, in order of both
invasiveness and performance impact, from smallest to largest.
1) Asterisk 1.6.1 introduces some additional logic to be able to handle
distributed device state. This functionality comes at a cost.
One relatively minor change in this patch is that the extra processing
required for distributed device state is now completely bypassed if
it's not needed.
2) One of the things that I noticed when profiling this code was that a
_lot_ of time was spent doing string comparisons. I changed the way
strings are represented in an event to include a hash value at the front.
So, before doing a string comparison, we do an integer comparison on the
hash.
3) Finally, the code that handles the event cache has been re-written.
I tried to do this in a such a way that it had minimal impact on the API.
I did have to change one API call, though - ast_event_queue_and_cache().
However, the way it works now is nicer, IMO. Each type of event that
can be cached (MWI, device state) has its own hash table and rules for
hashing and comparing objects. This by far made the biggest impact on
performance.
For additional details regarding this code and how it was tested, please see the
review request.
(closes issue #14738)
Reported by: russell
Review: http://reviewboard.digium.com/r/205/
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(closes issue #14714)
Reported by: ygor
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184147 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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functions.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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