https://origsvn.digium.com/svn/asterisk/branches/1.4
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r229360 | tilghman | 2009-11-10 16:09:16 -0600 (Tue, 10 Nov 2009) | 12 lines
If two pattern classes start with the same digit and have the same number of characters, they will compare equal.
The example given in the issue report is that of [234] and [246], which have
these characteristics, yet they are clearly not equivalent. The code still
uses these two characteristics, yet when the two scores compare equal, an
additional check will be done to compare all characters within the class to
verify equality.
(closes issue #15421)
Reported by: jsmith
Patches:
20091109__issue15421__2.diff.txt uploaded by tilghman (license 14)
Tested by: jsmith, thedavidfactor
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Two examples of its use are included, and the usage could be expanded in some
cases into certain configuration options where time periods are specified.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224225 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Channels are stored in an ao2_container. When accessing an item within
an ao2_container the proper locking order is to first lock the container,
and then the items within it.
In ast_do_masquerade both the clone and original channel must be locked
for the entire duration of the function. The problem with this is that
it attemptes to unlink and link these channels back into the ao2_container
when one of the channel's name changes. This is invalid locking order as
the process of unlinking and linking will lock the ao2_container while
the channels are locked!!! Now, both the channels in do_masquerade are
unlinked from the ao2_container and then locked for the entire function.
At the end of the function both channels are unlocked and linked back
into the container with their new names as hash values.
This new method of requiring all channels and tech pvts to be unlocked
before ast_do_masquerade() or ast_change_name() required several
changes throughout the code base.
(closes issue #15911)
Reported by: russell
Patches:
masq_deadlock_trunk.diff uploaded by dvossel (license 671)
Tested by: dvossel, atis
(closes issue #15618)
Reported by: lmsteffan
Patches:
deadlock_local_attended_transfers_trunk.diff uploaded by dvossel (license 671)
Tested by: lmsteffan, dvossel
Review: https://reviewboard.asterisk.org/r/387/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222761 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r218867 | dbrooks | 2009-09-16 13:00:45 -0500 (Wed, 16 Sep 2009) | 13 lines
Fixes CID pattern matching behavior to mirror that of extension pattern matching.
Pattern matching for extensions uses a type of scoring system, giving values for
specificity to each character in the pattern. Unfortunately, this is done character
by character, in order. This does lead to some less specific patterns being first
in line for matching, but it will usually get the job done.
This patch merely brings CID matching to the same level as extension matching.
This patch does not attempt to tackle the problem shared by extension matching.
(closes issue #14708)
Reported by: klaus3000
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r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines
Make apps send PROGRESS control frame for early media and fix too early media issue in SIP
The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI
links *before* any call progress. The SIP channel receives these frames and by default
signals 183 Session progress and starts sending media. This will cause phones to
play silence and ignore the later 180 ringing message. A bad user experience.
The fix is twofold:
- We discovered that asterisk apps that support early media ("noanswer") did not send
any PROGRESS frame to indicate early media. Fixed.
- We introduce a setting in chan_sip so that users can disable any relay of media frames
before the outbound channel actually indicates any sort of call progress.
In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions
of Asterisk, this will be enabled. We don't assume that it will change your Asterisk
phone experience - only for the better.
We encourage third-party application developers to make sure that if they have applications
that wants to send early media, add a PROGRESS control frame transmission to make sure that
all channel drivers actually will start sending early media. This has not been the default
in Asterisk previous to this patch, so if you got inspiration from our code, you need to
update accordingly. Sorry for the trouble and thanks for your support.
This code has been running for a few months in a large scale installation (over 250
servers with PRI and/or BRI links to old PBX systems).
That's no proof that this is an excellent patch, but, well, it's tested :-)
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r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009) | 14 lines
Improved mapping of extension states from combined device states.
This fixes a few issues with incorrect extension states and adds
a cli command, core show device2extenstate, to display all possible
state mappings.
(closes issue #15413)
Reported by: legart
Patches:
exten_helper.diff uploaded by dvossel (license 671)
Tested by: dvossel, legart, amilcar
Review: https://reviewboard.asterisk.org/r/301/
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CEL is the new system for logging channel events. This was inspired after
facing many problems trying to represent what is possible to happen to a call
in Asterisk using CDR records. For more information on CEL, see the built in
HTML or PDF documentation generated from the files in doc/tex/.
Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
work developing this code. Also, thanks to Matt Nicholson (mnicholson) and
Sean Bright (seanbright) for their assistance in the final push to get this
code ready for Asterisk trunk.
Review: https://reviewboard.asterisk.org/r/239/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In the definition of new_find_extension(), the arguments 'callerid' and
'label' were swapped. The prototype declaration and all calls to the
function are ordered 'callerid' then 'label', but the function itself
was ordered 'label' then 'callerid'.
(closes issue #15303)
Reported by: JimDickenson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199957 65c4cc65-6c06-0410-ace0-fbb531ad65f3
'channel originate ... application <app>' CLI command.
(And yeah, I cleaned up some whitespace in res_clioriginate.c... big whoop,
wanna fight about it!?)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196758 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A new xml element was created to manage the AMI actions documentation,
using AstXML.
To register a manager action using XML documentation it is now possible
using ast_manager_register_xml().
The CLI command 'manager show command' can be used to show the parsed
documentation.
Example manager xml documentation:
<manager name="ami action name" language="en_US">
<synopsis>
AMI action synopsis.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(...)" /> <-- for ActionID
<parameter name="header1" required="true">
<para>Description</para>
</parameter>
...
</syntax>
<description>
<para>AMI action description</para>
</description>
<see-also>
...
</see-also>
</manager>
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes:
- CLI command handlers
- CLI command handler arguments
- AGI command handlers
- AGI command handler arguments
- Dialplan application handler arguments
- Speech engine API function arguments
In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing.
Review: https://reviewboard.asterisk.org/r/251/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Update applications documentation to warn the user about the use of the
WaitExten() application within a Macro(). Recommend the use of Read()
instead.
(closes issue #14444)
Reported by: ewieling
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195162 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Always free the allocated memory for a string field, because
we are always using it (not only when xmldocs are enabled).
Also if there is an error allocating memory for the string field
remember to unlock the list of registered applications, before returning.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194945 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r194137 | tilghman | 2009-05-12 19:52:03 -0500 (Tue, 12 May 2009) | 7 lines
Fix logic for how to proceed with a single digit extension.
(closes issue #15091)
Reported by: andrew
Patches:
20090512__issue15091.diff.txt uploaded by tilghman (license 14)
Tested by: andrew
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1) If MALLOC_DEBUG is used when concurrently using ast_backtrace, the free()
used in that routine will trigger an error, because the memory was allocated
internally to libc, where we could not intercept that call to wrap it.
Therefore, it's not memory we knew about, and the free is reported as an
error.
2) Now that channels are objects, the old hack of initializing a channel
to all zeroes no longer works, since we may try to call something like
ast_channel_lock() within a function on that reference. In that case, it's
reported as an error, because the pointer isn't an object reference.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r193119 | tilghman | 2009-05-07 18:41:11 -0500 (Thu, 07 May 2009) | 19 lines
Fix Background within a Macro for FreePBX.
If the single digit DTMF is an extension in the specified context, then
go there and signal no DTMF. Otherwise, we should exit with that DTMF.
If we're in Macro, we'll exit and seek that DTMF as the beginning of an
extension in the Macro's calling context. If we're not in Macro, then
we'll simply seek that extension in the calling context. Previously,
someone complained about the behavior as it related to the interior of a
Gosub routine, and the fix (#14011) inadvertently broke FreePBX
(#14940). This change should fix both of these situations, but with the
possible incompatibility that if a single digit extension does not exist
(but a longer extension COULD have matched), it would have previously
gone immediately to the "i" extension, but will now need to wait for a
timeout.
(closes issue #14940)
Reported by: p_lindheimer
Patches:
20090420__bug14940.diff.txt uploaded by tilghman (license 14)
Tested by: p_lindheimer
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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
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
1. Differentiate between literal characters in an extension
and characters that should be treated as a pattern match. Prior to
these fixes, an extension such as NNN would be treated as a pattern,
rather than a literal string of N's.
2. Fixed the logic used when matching an extension with a bracketed
expression, such as 2[5-7]6.
3. Removed all areas of code that were executed when NOT_NOW was
#defined. The code in these areas had the potential to crash, for
one thing, and the actual intent of these blocks seemed counterproductive.
4. Fixed many many coding guidelines problems I encountered while looking
through the corresponding code.
5. Added failure cases and warning messages for when duplicate extensions
are encountered.
6. Miscellaneous fixes to incorrect or redundant statements.
(closes issue #14615)
Reported by: steinwej
Tested by: mmichelson
Review: http://reviewboard.digium.com/r/194/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188901 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r177786 | tilghman | 2009-02-20 16:59:52 -0600 (Fri, 20 Feb 2009) | 9 lines
Don't print the CR-NL combination when we aren't outputting to the manager.
An embedded CR-NL in a CLI command screws up several AMI parsers that don't
expect to see that combination in the middle of output.
(Closes issue #14305)
Reported by: martins
Patch by: tilghman
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Reason: when I re-engineered the merge_and_delete func to
reduce its lock time, I failed to notice that the
functions it calls still also do locking as before.
This leads to deadlocks on dialplan reloads, when
there are actually living, subscribed hints registered
in the system.
While the reporter come across this problem while using
AEL, I might note that these deadlocks should also happen
if extensions.conf were used.
Here I added these routines to pbx.c:
ast_add_extension_nolock
add_pri_lockopt
ast_add_extension2_lockopt
find_context
add_hint_nolock
All of the above routines are static and restricted
to be used only within pbx.c, and more specifically
within the merge_contexts_and_delete routine.
They are pretty much the same as their counterparts
except they don't lock contexts or hints.
Most of them now do the real work of their
name-alike, with optional locking via extra arguments,
and are called by their name-alike. The goal was to
have the original functions so they would behave
exactly as before.
Both PJ and I tested these fixes, and the deadlocking
problem is no longer encountered.
(closes issue #14357)
Reported by: pj
Patches:
14357.diff uploaded by murf (license 17)
Tested by: pj, murf
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176943 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The big problem here is that the 3rd argument provided in these uses of strncpy()
did not reserve a byte for the null terminator, leaving the potential for writing
one byte past the end of the buffer.
Aside from this, there were coding guidelines violations with regards to spacing,
as well as hard coded lengths being used instead of sizeof().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176901 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch includes a number of changes to the indications API. The primary
motivation for this work was to improve stability. The object management
in this API was significantly flawed, and a number of trivial situations could
cause crashes.
The changes included are:
1) Remove the module res_indications. This included the critical functionality
that actually loaded the indications configuration. I have seen many people
have Asterisk problems because they accidentally did not have an
indications.conf present and loaded. Now, this code is in the core,
and Asterisk will fail to start without indications configuration.
There was one part of res_indications, the dialplan applications, which did
belong in a module, and have been moved to a new module, app_playtones.
2) Object management has been significantly changed. Tone zones are now
managed using astobj2, and it is no longer possible to crash Asterisk by
issuing a reload that destroys tone zones while they are in use.
3) The API documentation has been filled out.
4) The API has been updated to follow our naming conventions.
5) Various bits of code throughout the tree have been updated to account
for the API update.
6) Configuration parsing has been mostly re-written.
7) "Code cleanup"
The code is from svn/asterisk/team/russell/indications/.
Review: http://reviewboard.digium.com/r/149/
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