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
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r195881 | mnicholson | 2009-05-21 10:25:50 -0500 (Thu, 21 May 2009) | 13 lines
This commit prevents cdr records with AST_CDR_FLAG_ANSLOCKED and AST_CDR_FLAG_LOCKED from being updated in certain cases.
This is accomplished by adding two functions to update the answer time and disposition of calls that checks for the proper lock flags. These functions are used in the ast_bridge_call() function so that ForkCDR(A) calls are respected.
This patch also modifies the way ast_bridge_call() chooses the cdr record to base the bridged_cdr on. Previously the first unlocked cdr record would be chosen, now instead the first cdr record is chosen and forked cdr records are moved to the bridge_cdr. This allows the original cdr record and any forked cdr records to be properly updated with answer and end times.
(closes issue #13797)
Reported by: sh0t
Tested by: sh0t
(closes issue #14744)
Reported by: deepesh
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195882 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The CLI command 'manager show command' supports passing multiple action names in
the same line, but it was not allowing that because of a incorrect check in the
argumentes counter. Also the documentation was updated to show that this usage
of the command is possible.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195369 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The CLI command 'manager show command' supports passing multiple AMI actions
at a time. The issue with this command was in another place.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When running the autocomplete of the CLI command 'manager show command <action>'
it was autocompleting everything else after the <action> argument, giving an error,
because this command doesn't support multiple AMI action names at a time.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195367 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This avoids possible conflicts with the internal implementation of
daemon(3).
(closes issue #15093)
Reported by: tzafrir
Patches:
20090513__issue15093__2.diff.txt uploaded by tilghman (license 14)
Tested by: tzafrir
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195320 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r195206 | file | 2009-05-18 12:51:22 -0300 (Mon, 18 May 2009) | 7 lines
Fix a typo which caused loss of audio when using G729 in some scenarios with a smoother present.
(closes issue #15105)
Reported by: bamby
Patches:
process-vad-correctly.diff uploaded by bamby (license 430)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195207 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
Avoid duplicating xml documentation by allowing to include other parts of
the xml documentation using XInclude.
Example:
<xi:include xpointer="xpointer(/docs/function[@name='CHANNEL']/synopsis)" />
(Insert this line to include the synopsis of the CHANNEL function xml
documentation).
It is also possible to include documentation from other files in the
'documentation/' directory using the href="" attribute inside a xinclude
element.
(closes issue #15107)
Reported by: lmadsen
(issue #14444)
Reported by: ewieling
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194982 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
It was not possible to use an enumlist inside an enum:
<enumlist>
<enum name="aa">
<enumlist>
...
</enumlist>
</enum>
</enumlist>
Now we will be able to insert as many levels as we want.
(closes issue #15112)
Reported by: lmadsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the ability for modules to dynamically create logger levels for their own use; these are named levels just like the built-in levels, and can be directed to any destination that the logger can send any level to, by including their names in logger.conf.
Review: https://reviewboard.asterisk.org/r/244/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194610 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There was a typo in the structure being locked, and we were locking the
'sessions' container instead of the 'session' structure thar we are modifying.
Reported by seanbright on #asterisk-dev, thanks!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194283 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194101 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the user logout the session was being destroyed twice and the file
descriptor was being closed twice. The sessions reference counter wasn't
used in a proper way.
The 'mansession' structure was being treated as an astobj2 and we were
calling ao2_lock/ao2_unlock causing astobj2 report a warning message and
not locking the structure.
Also we were using an ugly naming convention 'destroy_session',
'session_destroy', 'free_session', ... all this "duplicated" code was merged.
(closes issue #14974)
Reported by: pj
Patches:
manager.diff2 uploaded by eliel (license 64)
Tested by: dhubbard, eliel, mnicholson
(closes issue #15088)
Reported by: eliel
Review: http://reviewboard.asterisk.org/r/248/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r193193 | kpfleming | 2009-05-08 09:03:28 -0500 (Fri, 08 May 2009) | 7 lines
Make absolute paths for logger channels work properly
(Note: This is not a new feature, it was previously undocumented and broken.)
The Asterisk logger has a feature to support absolute pathnames for logger channels, but the code implementing the feature was broken. This has been fixed, and the absolute path feature is now documented in the sample logger.conf.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193194 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r192858 | jpeeler | 2009-05-06 17:15:19 -0500 (Wed, 06 May 2009) | 10 lines
Make ParkedCall application stop execution of the dialplan after hang up
Just changed park_exec to always return non-zero. I really wasn't entirely sure
at first if this was a bug. Decided it was since it would be surprising when
not using ParkedCall in the dialplan to hang up and have dialplan execution
continue.
(closes issue #14555)
Reported by: francesco_r
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The local struct ast_party_connected_line connected_caller variable
was uninitialized when the copy function was called.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192590 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r192454 | file | 2009-05-05 15:22:27 -0300 (Tue, 05 May 2009) | 8 lines
Fix an incorrect assumption that certain values on the channel will always exist when they may not.
The CDR code involved with bridges wrongly assumed that the currently executing application and data
values will always exist. It is possible for this to be false when call forwarding is involved.
(closes issue #14984)
Reported by: gincantalupo
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192462 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit adds an API call that can be used to allocate a structure along with this stringfield storage in a single allocation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192362 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
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
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191919 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- convert handling of IE PLTYPEs to switch statements
- add braces to various small blocks
- remove a bit of trailing whitespace
- remove a couple of unnecessary ast_strdupa() uses
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191848 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
This code was copy-and-pasted without properly changing references to event_rotate into queue_rotate, so under some conditions the log rotation would rotate queue_log even though it was not necessary.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191775 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r191488 | jpeeler | 2009-05-01 12:40:46 -0500 (Fri, 01 May 2009) | 9 lines
Fix DTMF not being sent to other side after a partial feature match
This fixes a regression from commit 176701. The issue was that
ast_generic_bridge never exited after the feature digit timeout had elapsed,
which prevented the queued DTMF from being sent to the other side.
This issue was reported to me directly.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If backgrounding and no core will be produced, then changing the directory
won't break anything; likewise, if the CWD isn't accessible by the current
user, then a core wasn't possible anyway.
(closes issue #14831)
Reported by: chris-mac
Patches:
20090428__bug14831.diff.txt uploaded by tilghman (license 14)
20090430__bug14831.diff.txt uploaded by tilghman (license 14)
Tested by: chris-mac
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191283 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191177 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
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
This commit stops a warning message (user_data is NULL) from getting output when
manager events get sent before manager is initialized. This happens because manager
is initialized *after* modules are loaded and the act of loading modules triggers
manager events.
(issue #14974)
Reported by: pj
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190586 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190545 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
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190057 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r189391 | dbailey | 2009-04-20 14:10:56 -0500 (Mon, 20 Apr 2009) | 4 lines
Clean up problem with manager implementation of mmap where it was not testing against MAP_FAILED response.
Got rid of shadowed variable used in processign the mmap results.
Change test of mmap results to compare against MAP_FAILED
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189419 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r189277 | mmichelson | 2009-04-20 09:04:41 -0500 (Mon, 20 Apr 2009) | 12 lines
Move the check for chan->fdno == -1 to after the zombie/hangup check.
Many users were finding that their hung up channels were staying up and
causing 100% CPU usage.
(issue #14723)
Reported by: seadweller
Patches:
14723_1-4-tip.patch uploaded by mmichelson (license 60)
Tested by: falves11, bamby
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189278 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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187491 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
(inspired by this evening's Toronto Asterisk Users Group meeting and previous dicussions amongst various community members)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187269 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r186984 | mmichelson | 2009-04-08 10:26:46 -0500 (Wed, 08 Apr 2009) | 24 lines
Make a couple of changes with regards to a new message printed in ast_read().
"ast_read() called with no recorded file descriptor" is a new message added
after a bug was discovered. Unfortunately, it seems there are a bunch of places
that potentially make such calls to ast_read() and trigger this error message
to be displayed. This commit does two things to help to make this message appear
less.
First, the message has been downgraded to a debug level message if dev mode is
not enabled. The message means a lot more to developers than it does to end users,
and so developers should take an effort to be sure to call ast_read only when
a channel is ready to be read from. However, since this doesn't actually cause an
error in operation and is not something a user can easily fix, we should not spam
their console with these messages.
Second, the message has been moved to after the check for any pending masquerades.
ast_read() being called with no recorded file descriptor should not interfere with
a masquerade taking place.
This could be seen as a simple way of resolving issue #14723. However, I still want
to try to clear out the existing ways of triggering this message, since I feel that
would be a better resolution for the issue.
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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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186953 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r186832 | mmichelson | 2009-04-07 18:49:49 -0500 (Tue, 07 Apr 2009) | 8 lines
Set the AST_FEATURE_WARNING_ACTIVE flag when a p2p bridge returns AST_BRIDGE_RETRY.
Without this flag set, warning sounds will not be properly played to either party
of the bridge.
(closes issue #14845)
Reported by: adomjan
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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186382 65c4cc65-6c06-0410-ace0-fbb531ad65f3
audio_audiohook_write_list() did not take into account that the sample size may change after translation depending on if the original frame is is 8khz or 16khz. the sample size is now updated after translating to reflect this possibility. This caused the audio on the receiving end to sound terrible. Thanks to jcolp and mmichelson for helping me work this out.
(issue AST-197)
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r185771 | russell | 2009-04-01 08:47:30 -0500 (Wed, 01 Apr 2009) | 6 lines
Fix a case where DTMF could bypass audiohooks.
This change fixes a situation where an audiohook that wants DTMF would not
actually get it. This is in the code path where we end DTMF digit length
emulation while handling a NULL frame.
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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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r185196 | file | 2009-03-31 11:06:39 -0300 (Tue, 31 Mar 2009) | 8 lines
Fix crash when moving audiohooks between channels.
Handle the scenario where we are called to move audiohooks between channels
and the source channel does not actually have any on it.
(closes issue #14734)
Reported by: corruptor
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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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184762 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r184188 | eliel | 2009-03-25 10:12:54 -0400 (Wed, 25 Mar 2009) | 13 lines
Avoid destroying the CLI line when moving the cursor backward and trying to autocomplete.
When moving the cursor backward and pressing TAB to autocomplete, a NULL is put
in the line and we are loosing what we have already wrote after the actual
cursor position.
(closes issue #14373)
Reported by: eliel
Patches:
asterisk.c.patch uploaded by eliel (license 64)
Tested by: lmadsen
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r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009) | 6 lines
Cleaning up a few things in detect disconnect patch
Initialized ast_call_feature in detect_disconnect to avoid accessing uninitialized memory. Cleaned up /param tags in features.h. No longer send dynamic features in ast_feature_detect.
issue #11583
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r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) | 17 lines
Allow disconnect feature before a call is bridged
feature.conf has a disconnect option. By default this option is set to '*', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c.
(closes issue #11583)
Reported by: sobomax
Patches:
patch-apps__app_dial.c uploaded by sobomax (license 359)
11583.latest-patch uploaded by murf (license 17)
detect_disconnect.diff uploaded by dvossel (license 671)
Tested by: sobomax, dvossel
Review: http://reviewboard.digium.com/r/195/
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If two channels were bridged together using a generic bridge the T38
control frame would get passed up instead of being indicated on the
other channel.
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r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines
Fix cases where the internal poll() was not being used when it needed to be.
We have seen a number of problems caused by poll() not working properly on
Mac OSX. If you search around, you'll find a number of references to using
select() instead of poll() to work around these issues. In Asterisk, we've
had poll.c which implements poll() using select() internally. However, we
were still getting reports of problems.
vadim investigated a bit and realized that at least on his system, even
though we were compiling in poll.o, the system poll() was still being used.
So, the primary purpose of this patch is to ensure that we're using the
internal poll() when we want it to be used.
The changes are:
1) Remove logic for when internal poll should be used from the Makefile.
Instead, put it in the configure script. The logic in the configure
script is the same as it was in the Makefile. Ideally, we would have
a functionality test for the problem, but that's not actually possible,
since we would have to be able to run an application on the _target_
system to test poll() behavior.
2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT
is not defined.
3) Change uses of poll() throughout the source tree to ast_poll(). I feel
that it is good practice to give the API call a new name when we are
changing its behavior and not using the system version directly in all cases.
So, normally, ast_poll() is just redefined to poll(). On systems where
AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll().
4) Change poll() in main/poll.c to be ast_internal_poll().
It's worth noting that any code that still uses poll() directly will work fine
(if they worked fine before). So, for example, out of tree modules that are
using poll() will not stop working or anything. However, for modules to work
properly on Mac OSX, ast_poll() needs to be used.
(closes issue #13404)
Reported by: agalbraith
Tested by: russell, vadim
http://reviewboard.digium.com/r/198/
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r182808 | kpfleming | 2009-03-17 20:55:22 -0500 (Tue, 17 Mar 2009) | 5 lines
Improve the build system to *properly* remove unnecessary symbols from the runtime global namespace. Along the way, change the prefixes on some internal-only API calls to use a common prefix.
With these changes, for a module to export symbols into the global namespace, it must have *both* the AST_MODFLAG_GLOBAL_SYMBOLS flag and a linker script that allows the linker to leave the symbols exposed in the module's .so file (see res_odbc.exports for an example).
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This does not change any behavior, but moves the frames from the local frame
list back to the channel read queue using an O(n) algorithm instead of O(n^2).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations.
When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames.
This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller.
http://reviewboard.digium.com/r/196/
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r182449 | tilghman | 2009-03-17 00:50:52 -0500 (Tue, 17 Mar 2009) | 7 lines
Fix race in astdb
The underlying db1 implementation does not fully isolate the pages retrieved
from astdb, so the lock protecting accesses needs to be extended until the
copy from the shared memory structure is done.
(closes issue #14682)
Reported by: makoto
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For a channel that is not yet answered this API call will wait
until a voice frame is received on the channel before returning.
It does this by waiting for frames on the channel and reading them
in. The frames read in were not freed when they should have been.
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r181990 | mmichelson | 2009-03-13 12:12:32 -0500 (Fri, 13 Mar 2009) | 35 lines
Check the DYNAMIC_FEATURES of both the chan and peer when interpreting DTMF.
Dynamic features defined in the applicationmap section of features.conf allow
one to specify whether the caller, callee, or both have the ability to use the
feature. The documentation in the features.conf.sample file could be interpreted
to mean that one only needs to set the DYNAMIC_FEATURES channel variable on the
calling channel in order to allow for the callee to be able to use the features
which he should have permission to use. However, the DYNAMIC_FEATURES variable
would only be read from the channel of the participant that pressed the DTMF
sequence to activate the feature. The result of this was that the callee was
unable to use dynamic features unless the dialplan writer had taken measures
to be sure that the DYNAMIC_FEATURES variable was set on the callee's channel.
This commit changes the behavior of ast_feature_interpret to concatenate the
values of DYNAMIC_FEATURES from both parties involved in the bridge. The features
themselves determine who has permission to use them, so there is no reason to believe
that one side of the bridge could gain the ability to perform an action that they
should not have the ability to perform.
Kevin Fleming pointed out on the asterisk-users list that the typical way that this
was worked around in the past was by setting _DYNAMIC_FEATURES on the calling channel
so that the value would be inherited by the called channel. While this works, the
documentation alone is not enough to figure out why this is necessary for the callee
to be able to use dynamic features. In this particular case, changing the code to match
the documentation is safe, easy, and will generally make things easier for people for
future installations.
This bug was originally reported on the asterisk-users list by David Ruggles.
(closes issue #14657)
Reported by: mmichelson
Patches:
14657.patch uploaded by mmichelson (license 60)
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Previously, only 5 columns were displayed, and if a translation time exceeded
99,999 useconds, it would be displayed as 0, instead of its actual time.
(closes issue #14532)
Reported by: pj
Patches:
20090311__bug14532.diff.txt uploaded by tilghman (license 14)
Tested by: pj
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r181423 | russell | 2009-03-11 16:42:58 -0500 (Wed, 11 Mar 2009) | 9 lines
Make code that updates BRIDGEPEER variable thread-safe.
It is not safe to read the name field of an ast_channel without the channel
locked. This patch fixes some places in channel.c where this was being done,
and lead to crashes related to masquerades.
(closes issue #14623)
Reported by: guillecabeza
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The main problem here was that cstdlib was undefining free thereby causing the
proper debug macros to not be used. ast_h323.cxx has been changed to call
ast_free instead to avoid the issue.
A few other issues were addressed:
- There were a few instances of functions improperly passing ast_free instead
of ast_free_ptr.
- Some clean up was done to avoid the debug macros intentionally being redefined.
(copied below from Kevin's commit, appreciate the help)
- disable astmm.h from doing anything when STANDALONE is defined, which is used
by the tools in the utils/ directory that use parts of Asterisk header files in
hackish ways; also ensure that utils/extconf.c and utils/conf2ael.c are
compiled with STANDALONE defined.
(closes issue #13593)
Reported by: pj
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181135 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Copied from my review board description:
This is a continuation of the API changes documentation started for describing
changes between releases. Most of the API changes were pretty simple needing
only to be brought to attention via the new "Asterisk API Changes" list.
However, if you see anything that needs further explanation feel free to
supplement what is there. The current method of documenting is to add (in the
header file): \version <ver number> <description of changes> and then to add
the function to the change list in doxyref.h on the AstAPIChanges page. I also
made sure all the functions that were newly added were tagged with \since
1.6.1. I think this is a good habit to start both for the historical aspect as
well as for the future ability to easily add a "New Asterisk API" page.
Review: http://reviewboard.digium.com/r/190/
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r180532 | dvossel | 2009-03-06 11:19:55 -0600 (Fri, 06 Mar 2009) | 9 lines
Fix handling of backreferences for ENUM lookups
enum.c did not handle regex backtraces correctly. The '\1' in the regex is a backreference that requires a pattern match to be inserted. The way the code used to work is that it would find the backreference and insert the entire input string minus the '+'. This is incorrect. The regexec() function takes in a variable called pmatch which is an array of structs containing the start and end indexes for each backreference substring. The original code actually passed the pmatch array pointer into regexec but never did anything with it. Now when a backtrace is found, the backtrace number is looked up in the pmatch array and the correct substring is inserted.
(closes issue #14576)
Reported by: chris-mac
Review: http://reviewboard.digium.com/r/187/
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r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar 2009) | 9 lines
Fix problems when RTP packet frame size is changed
During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good.
This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes.
Review: http://reviewboard.digium.com/r/184/
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This commit brings in the bridging core, bridging technologies,
and the ConfBridge application.
For usage information on the ConfBridge application please see
the output of "core show application ConfBridge" from the CLI.
For API documentation please see the doxygen page describing the
architecture and the documentation for each API call.
Review: http://reviewboard.digium.com/r/93/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180369 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In app.c, ast_app_getdata is called to stream the prompts and receive DTMF input. If ast_app_getdata() receives an empty string caused by the user inputing the end of string character, in this case '#', it should break from the prompt loop and return to app_read, but instead it cycles through all the prompts. I've added a return value for this special case in ast_readstring() which uses an enum I've delcared in apps.h. This enum is now used as a return value for ast_app_getdata().
(closes issue #14279)
Reported by: Marquis
Patches:
fix_app_read.patch uploaded by Marquis (license 32)
read-ampersanmd.patch2 uploaded by dvossel (license 671)
Tested by: Marquis, dvossel
Review: http://reviewboard.digium.com/r/177/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180032 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
I had some work to do to port these changes to trunk; the
check_expr stuff hasn't been updated here for quite some
time, it appears. I added some more tests to the check_expr2
suite. I had to play around with the makefile a bit, etc.
I added STANDALONE2 #ifdefs to ast_expr2.y so as not to
conflict structure with aelparse.
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r179807 | murf | 2009-03-03 11:11:34 -0700 (Tue, 03 Mar 2009) | 19 lines
These changes allow AEL to better check ${} constructs within $[...], that are concatenated with text.
I modified and added rules in ast_expr2.fl to better handle
the concatenations.
I added some default routines to ast_expr2.y so the standalone would
compile. It also looks like I haven't run this thru bison since 2.1, so
it's good to get this updated.
The Makefile has comments added now for check_expr2 and check_expr to
explain what they are for, and how to run them.
The testexpr2s stuff has been removed, in favor of check_expr2.
expr2.testinput has been updated to include the two expressions
that inspired these changes (from mcnobody on #asterisk this morning)
The regression has been run and all looks well.
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r179840 | file | 2009-03-03 14:27:09 -0400 (Tue, 03 Mar 2009) | 9 lines
Do not assume that the bridge_cdr is still attached to the channel when the 'h' exten is finished executing.
It is possible for a masquerade operation to occur when the 'h' exten is operating. This operation moves
the CDR records around causing the bridge_cdr to no longer exist on the channel where it is expected to.
We can not safely modify it afterwards because of this, so don't even try.
(closes issue #14564)
Reported by: meric
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r179741 | russell | 2009-03-03 10:45:46 -0600 (Tue, 03 Mar 2009) | 6 lines
Ensure chan->fdno always gets reset to -1 after handling a channel fd event.
Since setting fdno to -1 had to be moved, a couple of other code paths that
do process an fd event return early and do not pass through the code path
where it was moved to. So, set it to -1 in a few other places, too.
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r179671 | file | 2009-03-03 10:38:09 -0400 (Tue, 03 Mar 2009) | 3 lines
Move where fdno is set to the default value to *after* the read callback of the channel driver is called.
We have to do this as the underlying channel driver may need the fdno value to determine what to read.
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r179608 | russell | 2009-03-03 07:53:52 -0600 (Tue, 03 Mar 2009) | 9 lines
Make it easier to detect an improper call to ast_read().
When you call ast_waitfor() on a channel, the index into the channel fds array
that holds the file descriptor that poll() determines has input available is
stored in fdno. This patch clears out this value after a call to ast_read()
and also reports errors if ast_read() is called without an fdno set.
From a discussion on the asterisk-dev list.
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r179536 | jpeeler | 2009-03-02 17:54:39 -0600 (Mon, 02 Mar 2009) | 15 lines
Fix bridging regression from commit 176701
This fixes a bad regression where the bridge would exit after an attended
transfer was made. The problem was due to nexteventts getting set after the
masquerade which caused the bridge to return AST_BRIDGE_COMPLETE.
(closes issue #14315)
Reported by: tim_ringenbach
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r179468 | tilghman | 2009-03-02 17:09:01 -0600 (Mon, 02 Mar 2009) | 10 lines
When ending a recording with silence detection, remember to reduce the duration.
The end of the recording is correspondingly trimmed, but the duration was not
trimmed by the number of seconds trimmed, so the saved duration was necessarily
longer than the actual soundfile duration.
(closes issue #14406)
Reported by: sasargen
Patches:
20090226__bug14406.diff.txt uploaded by tilghman (license 14)
Tested by: sasargen
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r179461 | russell | 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009) | 8 lines
Ensure that only one thread is calling ast_settimeout() on a channel at a time.
For example, with an IAX2 channel, you can have both the channel thread and the
chan_iax2 processing threads calling this function, and doing so twice at the
same time is a bad thing.
(Found in a debugging session with dvossel and mmichelson)
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In this case, it's just a matter of reducing the default timeouts from 2000
to 1000 msec, as the max def feature digit timeout is no longer halved.
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r178956 | murf | 2009-02-26 14:27:32 -0700 (Thu, 26 Feb 2009) | 18 lines
This change moves the default feature digit timeout to 1000 ms from the previous default of 500.
As per bug 14515, a dev discussion arrived at a "mediated concensus"
of a default feature digit timeout of 1.0 sec. Some voted for 1300;
ctooley thought 1500 for distracted phone users in phone booths;
kpfleming put his foot down at 1.0 sec.
Users who found the previous default max delay of 250 msec perfect,
are welcome to override the new default. Notice that I said that
250 msec was the default; wait a minute, you might say, the config
file said it was 500 msec!; well, because of the bug fix for 14515,
we found that 500 msec was actually enforcing a max of 250. The bug
fix would restore 500 msec, but we felt even that was a bit tight
for most users... 2000 msec was pushed earlier by mmichelson, so
that reduces to 1000 msec after the bug fix. Enjoy!
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r178804 | murf | 2009-02-26 10:09:03 -0700 (Thu, 26 Feb 2009) | 28 lines
This patch prevents the feature detection timeout from being cut in half.
Because the ast_channel_bridge() call will return 0 and pass
a frame pointer for both DTMF_BEGIN and DTMF_END, the feature_timer
field in hte config struct is getting decremented twice, which
effectively cuts the digittimeout in half. I added conditions
to the if statement to only let DTMF_END frames to flow thru,
which solved the problem. Also, when the frame pointer is null,
let control flow thru-- this usually happens on timeouts. I added
a comment to the code to explain what's going on and why.
Many thanks to sodom for reporting this problem. Personnally, it always seemed
like something was wrong with the featuredigittimeout, but I never
could quite decide what... and was too busy to investigate.
This bug forced the issue, and now we know.
Sodom had other issues in 14515, but I couldn't reproduce them. If
he still has problems, and wants to get them solved, he is welcome
to reopen 14515.
(closes issue #14515)
Reported by: sodom
Patches:
14515.patch uploaded by murf (license 17)
Tested by: murf, sodom
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r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009) | 14 lines
Fix infinite DTMF when a BEGIN is received without an END.
This commit is related to rev 175124 of 1.4 where a previous attempt was made
to fix this problem. The problem with the previous patch was that the inserted
code needed to go _before_ setting the lastrxts to the current timestamp.
Because those were the same, the dtmfcount variable was never decremented, and
so the END was never sent.
In passing, I removed the dtmfsamples variable which was completed unused. I
also removed a redundant setting of the lastrxts variable.
(closes issue #14460)
Reported by: moliveras
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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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Trunk was already pretty 8-bit clean; but I'm still
removing the --full from the flex command so everything
is uniform.
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r177540 | murf | 2009-02-19 15:51:37 -0700 (Thu, 19 Feb 2009) | 21 lines
This patch fixes a problem with 8-bit input to the ast_expr2 scanner.
The real culprit was the --full argument to flex
in the Makefile! This causes a 7-bit scanner to be
generated.
I reviewed the rules and found one rule where I needed
to specifically include 8-bit chars for a token.
I tested against the text supplied by ibercom, and
all looks very well.
This has been there a surprisingly long time!
(closes issue #14498)
Reported by: ibercom
Patches:
14498.patch uploaded by murf (license 17)
Tested by: murf
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action_bridge() and bridge_exec() both search for the channels to bridge to, and then immediately drop the lock. Instead, they should hold the lock until the masquerade is complete. This will guarantee the channel remains and prevent any other weirdness from occurring. In action_bridge() some more weirdness comes into play. Both channels are needlessly locked at the same time and perform the exact same logic. It makes sense from a coding organizational standpoint, but could cause a theoretical deadlock so I split the code up. There is an issue associated with this, but since its a rather complicated thing to reproduce I'm not certain this alone will close it.
issue# 14296
Review: http://reviewboard.digium.com/r/167/
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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().
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r176701 | jpeeler | 2009-02-17 15:54:34 -0600 (Tue, 17 Feb 2009) | 17 lines
Modify bridging to properly evaluate DTMF after first warning is played
The main problem is currently if the Dial flag L is used with a warning sound,
DTMF is not evaluated after the first warning sound. To fix this, a flag has
been added in ast_generic_bridge for playing the warning which ensures that if
a scheduled warning is missed, multiple warrnings are not played back (due to a
feature evaluation or waiting for digits). ast_channel_bridge was modified to
store the nexteventts in the ast_bridge_config structure as that information
was lost every time ast_channel_bridge was reentered, causing a hangup due to
incorrect time calculations.
(closes issue #14315)
Reported by: tim_ringenbach
Reviewed on reviewboard:
http://reviewboard.digium.com/r/163/
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1) Add module use count handling so that timing modules can be unloaded.
2) Implement unload_module() functions for the timing interface modules.
3) Allow multiple timing modules to be loaded, and use the one with the
highest priority value.
4) Report which timing module is being use in the "timing test" CLI command.
(closes issue #14489)
Reported by: russell
Review: http://reviewboard.digium.com/r/162/
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This patch changes the scheduler to use a max-heap to store pending scheduler
entries instead of a fully sorted doubly linked list. When the number of
entries in the scheduler gets large, this will perform much better. For much
more detailed information on this change, see the review request.
Review: http://reviewboard.digium.com/r/160/
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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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The problem here is that the hint processing code was subscribed to the wrong
event type. So, it started processing state for a hint too soon, before the
device state cache had been updated.
Also, fix a similar bug in app_queue, as it was also subscribed to the wrong
event type.
(closes issue #14461)
Reported by: alecdavis
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r176216 | kpfleming | 2009-02-16 15:10:38 -0600 (Mon, 16 Feb 2009) | 3 lines
fix a flaw in the ast_string_field_build() family of API calls; these functions made no attempt to reuse the space already allocated to a field, so every time the field was written it would allocate new space, leading to what appeared to be a memory leak.
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r176254 | kpfleming | 2009-02-16 15:41:46 -0600 (Mon, 16 Feb 2009) | 3 lines
correct a logic error in the last stringfields commit... don't mark additional space as allocated if the string was built using already-allocated space
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I was finding that on my dev box, occasionally attempting to "stop now" in
trunk would cause Asterisk to hang. I traced this to the fact that the logger
thread was waiting on a condition which had already been signalled. The logger
thread also need to be sure to check the value of the close_logger_thread variable.
The close_logger_thread variable is only checked when the list of logmessages is empty.
This allows for the logger thread to print and free any pending messages before exiting.
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1) It had numerous coding guidelines violations with regards to formatting.
2) It allocated memory using ast_calloc() that was never freed.
3) It didn't check for failure from the allocation.
4) It used sprintf() and strcat() to build the result, doing zero checking to
prevent writing past the end of the provided buffer.
The function also lacks API documentation, but that has not been addressed in
this commit.
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This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.
Along the way, some related work was done:
1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.
2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.
3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).
Review: http://reviewboard.digium.com/r/158/
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r175311 | tilghman | 2009-02-12 15:19:40 -0600 (Thu, 12 Feb 2009) | 9 lines
Fix crashes when receiving certain T.38 packets. Also, increase the maximum
size of T.38 packets and warn users when they try to set the limits above those
maximums.
(closes issue #13050)
Reported by: schern
Patches:
20090212__bug13050.diff.txt uploaded by Corydon76 (license 14)
Tested by: schern
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r175294 | jpeeler | 2009-02-12 14:34:36 -0600 (Thu, 12 Feb 2009) | 9 lines
Fix ParkedCall event information for From field in the case of a blind transfer
If the parker information can not be obtained from the peer, try and see if
the BLINDTRANSFER channel variable has been set. Previously, a blind transfer
to the ParkAndAnnounce app would return nothing for the From.
Closes AST-189
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r175187 | jpeeler | 2009-02-12 11:57:10 -0600 (Thu, 12 Feb 2009) | 6 lines
Fix crash in event of failed attempt to transfer to parking
The peer may not necessarily exist, such as in the case of a transfer to
ParkAndAnnounce. In this case don't try to play a sound to it.
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r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009) | 27 lines
Don't send DTMF for infinite time if we do not receive an END event.
I thought that this was going to end up being a pretty gnarly fix, but it turns
out that there was actually already a configuration option in rtp.conf,
dtmftimeout, that was intended to handle this situation. However, in between
Asterisk 1.2 and Asterisk 1.4, the code that processed the option got lost.
So, this commit brings it back to life.
The default timeout is 3 seconds. However, it is worth noting that having
this be configurable at all is not really the recommended behavior in RFC 2833.
From Section 3.5 of RFC 2833:
Limiting the time period of extending the tone is necessary
to avoid that a tone "gets stuck". Regardless of the
algorithm used, the tone SHOULD NOT be extended by more than
three packet interarrival times. A slight extension of tone
durations and shortening of pauses is generally harmless.
Three seconds will pretty much _always_ be far more than three packet
interarrival times. However, that behavior is not required, so I'm going to
leave it with our legacy behavior for now.
Code from svn/asterisk/team/russell/issue_14460
(closes issue #14460)
Reported by: moliveras
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Core show locks information involving an ao2_trylock did not
show the function that called ao2_trylock, but would instead
show ao2_trylock as the source of the lock. This is not useful
when trying to debug locking issues.
One bizarre note is that this logic is already in 1.4 but somehow
did not get merged to trunk or the 1.6.X branches.
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The 'd' option would not work for channel types which use RTP to transport
DTMF digits. The only way to allow for this to work was to answer the channel
if we saw that this option was enabled.
I realized that this may cause issues with CDRs, specifically with giving false
dispositions and answer times. I therefore modified ast_answer to take another
parameter which would tell if the CDR should be marked answered.
I also extended this to the Answer application so that the channel may be answered
but not CDRified if desired.
I also modified app_dictate and app_waitforsilence to only answer the channel if it
is not already up, to help not allow for faulty CDR answer times.
All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all
the changes except for the change to the Answer application will go in since we do
not introduce new features into stable branches
(closes issue #14164)
Reported by: DennisD
Patches:
14164.patch uploaded by putnopvut (license 60)
Tested by: putnopvut
Review: http://reviewboard.digium.com/r/145
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We need to do this because while we know that the freeing of the channel may cause something to become
not in use we do not know this for sure. There may be another channel that is still up which would cause
it to be in use.
(closes issue #13238)
Reported by: kowalma
Patches:
20090121__bug13238.diff.txt uploaded by Corydon76 (license 14)
Tested by: alecdavis
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The explanation behind this fix is a bit complicated, and I've already
typed it up in the code as a huge comment inside of manager.c, so I'll
give the abridged version here.
We needed a way to separate action-specific data from session-specific data.
Unfortunately, the only way to maintain API compatibility and to not have to
change every single manager action was to rename the current mansession structure
and wrap it inside a new mansession structure which actually contains action-
specific data.
(closes issue #14364)
Reported by: awk
Patches:
14364_better.patch uploaded by putnopvut (license 60)
Tested by: putnopvut
Review: http://reviewboard.digium.com/r/148/
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r174583 | mnicholson | 2009-02-10 11:52:42 -0600 (Tue, 10 Feb 2009) | 18 lines
Improve behavior of jitterbuffer when maxjitterbuffer is set.
This change improves the way the jitterbuffer handles maxjitterbuffer and
dramatically reduces the number of frames dropped when maxjitterbuffer is
exceeded. In the previous jitterbuffer, when maxjitterbuffer was exceeded, all
new frames were dropped until the jitterbuffer is empty. This change modifies
the code to only drop frames until maxjitterbuffer is no longer exceeded.
Also, previously when maxjitterbuffer was exceeded, dropped frames were not
tracked causing stats for dropped frames to be incorrect, this change also
addresses that problem.
(closes issue #14044)
Patches:
bug14044-1.diff uploaded by mnicholson (license 96)
Tested by: mnicholson
Review: http://reviewboard.digium.com/r/144/
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This commit expands the Asterisk scheduler API to include a common implementation
of a scheduler context being processed by a dedicated thread. chan_iax2 has been
updated to use this new code. Also, as a result, this resolves some race
conditions related to the previous chan_iax2 scheduler handling.
Related to rev 171452 which resolved the same issues in 1.4.
Code from team/russell/sched_thread2
Review: http://reviewboard.digium.com/r/129/
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r173211 | jpeeler | 2009-02-03 15:57:01 -0600 (Tue, 03 Feb 2009) | 17 lines
Parking attempts made to one end of a bridge no longer will hang up due to a
parking failure.
Parking attempts made using either one-touch, or doing either a blind or
assisted transfer to the parking extension now keep up the bridge instead of
hanging up the attempted parked party. Normal causes for the parking attempt
to fail includes the specific specified extension (via PARKINGEXTEN) not being
available or if all the parking spaces are currently in use. To avoid having
to reverse a masquerade park_space_reserve was made to provide foresight if
a parking attempt will succeed and if so reserve the parking space.
(closes issue #13494)
Reported by: mdu113
Reviewed by Russell: http://reviewboard.digium.com/r/133/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173500 65c4cc65-6c06-0410-ace0-fbb531ad65f3
an fseek() on the stream, which is an invalid operation for a socket. Turning
off buffering explicitly lets the stdio functions know they cannot do this,
thus avoiding a potential error.
(closes issue #14400)
Reported by: fnordian
Patches:
tcptls.patch uploaded by fnordian (license 110)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173458 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The problem came from the fact that a frame read from a format interpreter
was not freed. Adding a call to ast_frfree fixed this. The explanation for
why this caused the problem is a bit complex, but here goes:
There was a problem in all versions of Asterisk where the embedded frame
of a filestream structure was referenced after the filestream was freed. This
was fixed by adding reference counting to the filestream structure. The refcount
would increase every time that a filestream's frame pointer was pointing to an
actual frame of data. When the frame was freed, the refcount would decrease. Once
the refcount reached 0, the filestream was freed, and as part of the operation,
the open files were closed as well.
Thus it becomes more clear why a missing ast_frfree would cause a reference leak
and cause the files to not be closed. You may ask then if there was a frame leak
before this patch. The answer to that is actually no! The filestream code was
"smart" enough to know that since the frame we received came from a format interpreter,
the frame had no malloced data and thus didn't need to be freed. Now, however, there
is cleanup that needs to be done when we finish with the frame, so we do need to
call ast_frfree on the frame to be sure that the refcount for the filestream is
decremented appropriately.
(closes issue #14384)
Reported by: fiddur
Patches:
14384.patch uploaded by putnopvut (license 60)
Tested by: fiddur, putnopvut
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173354 65c4cc65-6c06-0410-ace0-fbb531ad65f3
interfere with correct parsing of the extension. Also, if an unterminated
character class DOES make its way into the pbx core (through some other
method), ensure that it does not crash Asterisk.
(closes issue #14362)
Reported by: Nick_Lewis
Patches:
20090129__bug14362.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2. Use curl to download sound files, as curl is installed natively on OS X,
whereas wget and fetch are not.
(closes issue #14332)
Reported by: oej
Tested by: Corydon76
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173130 65c4cc65-6c06-0410-ace0-fbb531ad65f3
to be used within the dial app, before a call is bridged.
Many thanks to sobomax for submitting this patch.
Quoting from bug 11582:
"So the goal of the patch was to use the user configured feature code during the
call setup phase. The original ast_feature_interpret() function is not well suited
for this purpose as it uses much call bridge specific data and doesn't separate a
detection of feature from a feature handler call. So a new function ast_feature_detect()
has been extracted off the ast_feature_interpret() function but keeping the original
logic intact except some insignificant changes to locking.
"Having created the ast_feature_detect() function the possibility to use feature detection
in almost any place of the asterisk code. So a call to this function has been added to
wait_for_answer() function of app_dial.so module. This code doesn't call the feature handler
however and uses old call leg disconnect logic to make the changes as small and simple as
possible to prevent unexpected problems. A disconnect feature currently is the only one
supported during call setup as other features as call parking and call transfer don't make much
sense during call setup. However if need in some of the features would arise it is much easier to
implement as the infrastructure changes are already in place with this patch."
I have cleaned up the patch somewhat, and verified that the existing functionality is not
harmed, and that the new functionality works. Terry has committed his stuff, and there were
no conflicts (see 14274).
(closes issue #11583)
Reported by: sobomax
Patches:
patch-apps__app_dial.c uploaded by sobomax (license 359)
patch-include__asterisk__features.h uploaded by sobomax (license 359)
patch-res__res_features.c uploaded by sobomax (license 359)
enable-features-during-call-setup.diff uploaded by sobomax (license 359)
11583.newdiff uploaded by murf (license 17)
enable-features-during-call-setup-1.diff uploaded by sobomax (license 359)
11583.latest-patch uploaded by murf (license 17)
Tested by: sobomax, murf
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172890 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) | 37 lines
Fix feature inheritance with builtin features
When using builtin features like parking and transfers, the AST_FEATURE_* flags
would not be set correctly for all instances when either performing a builtin
attended transfer, or parking a call and getting the timeout callback. Also,
there was no way on a per-call basis to specify what features someone should
have on picking up a parked call (since that doesn't involve the Dial() command).
There was a global option for setting whether or not all users who pickup a
parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT,
AUTOMON, or PARKCALL.
This patch:
1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the
dialplan or with setvar in channels that support it. This variable can be set
to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the
equivalent dial options), to set what features should be activated on this
channel. The patch moves the setting of the features datastores into the
bridging code instead of app_dial to help facilitate this.
2) adds global options parkedcallparking, parkedcallhangup, and
parkedcallrecording to be similar to the parkedcalltransfers option for
globally setting features.
3) has builtin_atxfer call builtin_parkcall if being transfered to the parking
extension since tracking everything through multiple masquerades, etc. is
difficult and error-prone
4) attempts to fix all cases of return calls from parking and completed builtin
transfers not having the correct permissions
(closes issue #14274)
Reported by: aragon
Patches:
fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396)
Tested by: aragon, otherwiseguy
Review http://reviewboard.digium.com/r/138/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r172438 | tilghman | 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009) | 9 lines
Lose the CAP_NET_ADMIN at every fork, instead of at startup. Otherwise, if
Asterisk runs as a non-root user and the administrator does a 'restart now',
Asterisk loses the ability to set QOS on packets.
(closes issue #14004)
Reported by: nemo
Patches:
20090105__bug14004.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172441 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Also, change a function in app.c to return a userful value instead of always returning 0.
Patch by fnordian, changed by Corydon76 and myself.
This does not close the bug report, as fnordian had an additional change we're still discussing.
(related to issue #14059)
Reported by: fnordian
Patches:
chan_sip_hfield.patch uploaded by fnordian (license 110)
20090116__bug14059.diff.txt uploaded by Corydon76 (license 14)
Tested by: fnordian, Corydon76, oej
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) | 46 lines
This patch fixes h-exten running misbehavior in manager-redirected
situations.
What it does:
1. A new Flag value is defined in include/asterisk/channel.h,
AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the
bridge hangup exten code not to run the h-exten there (nor
publish the bridge cdr there). It will done at the pbx-loop
level instead.
2. In the manager Redirect code, I set this flag on the channel
if the channel has a non-null pbx pointer. I did the same for the
second (chan2) channel, which gets run if name2 is set...
and the first succeeds.
3. I restored the ending of the cdr for the pbx loop h-exten
running code. Don't know why it was removed in the first place.
4. The first attempt at the fix for this bug was to place code
directly in the async_goto routine, which was called from a
large number of places, and could affect a large number of
cases, so I tested that fix against a fair number of transfer
scenarios, both with and without the patch. In the process,
I saw that putting the fix in async_goto seemed not to affect
any of the blind or attended scenarios, but still, I was
was highly concerned that some other scenarios I had not tested
might be negatively impacted, so I refined the patch to
its current scope, and jmls tested both. In the process, tho,
I saw that blind xfers in one situation, when the one-touch
blind-xfer feature is used by the peer, we got strange
h-exten behavior. So, I inserted code to swap CDRs and
to set the HANGUP_DONT field, to get uniform behavior.
5. I added code to the bridge to obey the HANGUP_DONT flag,
skipping both publishing the bridge CDR, and running
the h-exten; they will be done at the pbx-loop (higher)
level instead.
6. I removed all the debug logs from the patch before committing.
7. I moved the AUTOLOOP set/reset in the h-exten code in res_features
so it's only done if the h-exten is going to be run. A very
minor performance improvement, but technically correct.
(closes issue #14241)
Reported by: jmls
Patches:
14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17)
Tested by: murf, jmls
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172063 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r171621 | mmichelson | 2009-01-27 14:06:01 -0600 (Tue, 27 Jan 2009) | 18 lines
Prevent a crash from occurring when a jitter buffer interpolated frame is
removed from a slinfactory
slinfactory used the "samples" field of an ast_frame in order to determine
the amount of data contained within the frame. In certain cases, such as
jitter buffer interpolated frames, the frame would have a non-zero value for
"samples" but have NULL "data"
This caused a problem when a memcpy call in ast_slinfactory_read would attempt
to access invalid memory. The solution in use here is to never feed frames into
the slinfactory if they have NULL "data"
(closes issue #13116)
Reported by: aragon
Patches:
13116.diff uploaded by putnopvut (license 60)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171622 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r170648 | file | 2009-01-23 16:16:39 -0400 (Fri, 23 Jan 2009) | 4 lines
When a channel is answered make sure any indications currently playing stop. Usually the phone would do this but if the channel was already answered then they are being generated by Asterisk and we darn well need to stop them.
(closes issue #14249)
Reported by: RadicAlish
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170652 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r170392 | mmichelson | 2009-01-23 09:40:39 -0600 (Fri, 23 Jan 2009) | 28 lines
Fix broken call pickup
There was a subtle change in ast_do_masquerade which
resulted in failed attempts to pickup calls. The problem
was that the value of the AST_FLAG_OUTGOING flag was
copied from the clone to the original channel. In the case
of call pickup, this meant that the AST_FLAG_OUTGOING flag
ended up being cleared on the channel that was attempting
to execute the pickup.
Because this flag was not set, when ast_read came across
an answer frame, it ignored it. The result of this was that
the calling channel was never properly answered.
This fix changes the behavior in ast_do_masquerade to set
the flags on the original channel to the union of the flags
on the clone channel. This way, if the AST_FLAG_OUTGOING
flag is set on either of the two channels involved in the
masquerade, the resulting channel will have the flag set
as well.
(closes issue #14206)
Reported by: francesco_r
Patches:
14206.patch uploaded by putnopvut (license 60)
Tested by: francesco_r, aragon, putnopvut
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r170050 | file | 2009-01-22 11:13:56 -0400 (Thu, 22 Jan 2009) | 6 lines
Do a string comparison instead of pointer comparison since some people specify the context they are actually in as an argument to get around some funkiness.
(closes issue #14011)
Reported by: dveiga
Patches:
pbx.c.patch uploaded by dveiga (license 665)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170051 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit fixes a crash that was occurring when attempting to
say a number between 10000 and 100000 due to dividing by 0.
This also removes some places where a "zero" is spoken when it
should not be.
(closes issue #14291)
Reported by: dant
Patches:
say.c-14291.diff uploaded by dant (license 670)
Tested by: dant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169794 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r169722 | tilghman | 2009-01-21 15:02:32 -0600 (Wed, 21 Jan 2009) | 8 lines
Extra NULLs in the output cause some terminal types to abort in the middle of
a color code, causing terminal weirdness.
(closes issue #14130)
Reported by: coolmig
Patches:
20090121__bug14130.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, coolmig
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169723 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes a problem that caused chan_sip to think that every open TCP session
was to a remote address of 0.0.0.0:0.
(closes issue #14287)
Reported by: jamesgolovich
Patches:
bug-14287.diff.txt uploaded by jamesgolovich (license 176)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169620 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is an ugly hack from 1.4 that allows the timeout callback from a parked
call to use the right channel name for the callback when the park is done with
a builtin attended transfer (that isn't completed early). This hasn't ever
worked in trunk and no one has complained yet, so eh.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169510 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The HW_PHYSMEM64 is only available in latest OpenBSD and/or amd64 versions of OpenBSD.
Use HW_PHYSMEM when HW_PHYSMEM64 is not available.
(closes issue #14129)
Reported by: ys
Patches:
2009011600_physmem64.diff.txt uploaded by mvanbaak (license 7)
Tested by: mvanbaak, jtodd
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168716 | twilson | 2009-01-15 12:22:49 -0600 (Thu, 15 Jan 2009) | 12 lines
Convert call to park_call_full to masq_park_call_announce
Since we removed the AST_PBX_KEEPALIVE return value, we need to use masqueraded
parking, otherwise we will try to call ast_hangup() in __pbx_run() and in
do_parking_thread() and then promptly crash.
(closes issue #14215)
Reported by: waverly360
Tested by: otherwiseguy
(closes issue #14228)
Reported by: kobaz
Tested by: otherwiseguy
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168941 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If the user says count=ENUMLOOKUP(${EXTEN},ALL,c,,enum.mydomain.tld);
then it won't complain about the empty arg (c,,...) and fabled's patch
won't let it swap the commas for pipes.
Ran it thru my dialplan and no complaints.
(closes issue #14169)
Reported by: fabled
Patches:
function-argument-separator-fix.diff uploaded by fabled (license 448)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168737 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This started as work to fix the 'core show sysinfo'
CLI command but while working on it oej
pointed out that read_credentials did not compile neither.
So while being there, fix that as well.
Thanks for all the testing oej!
(closes issue #14129)
Reported by: ys
Tested by: oej, mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168603 | tilghman | 2009-01-14 13:02:55 -0600 (Wed, 14 Jan 2009) | 7 lines
Don't read into a buffer without first checking if a value is beyond the end.
(closes issue #13600)
Reported by: atis
Patches:
20090106__bug13600.diff.txt uploaded by Corydon76 (license 14)
Tested by: atis
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168604 65c4cc65-6c06-0410-ace0-fbb531ad65f3
at this level prior to a large patch merge which converted ast_verbose
calls to ast_verb
(closes issue #14221)
Reported by: jcovert
Patches:
srv.c.patch uploaded by jcovert (license 551)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168523 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
In order to merge this 1.4 patch into trunk,
I had to resolve some conflicts and wait for
Russell to make some changes to res_agi.
I re-ran all the tests; 39 calls in all, and
made fairly careful notes and comparisons: I
don't want this to blow up some aspect of
asterisk; I completely removed the KEEPALIVE
from the pbx.h decls. The first 3 scenarios
involving feature park; feature xfer to 700;
hookflash park to Park() app call all behave
the same, don't appear to leave hung channels,
and no crashes.
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r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) | 131 lines
This merges the masqpark branch into 1.4
These changes eliminate the need for (and use of)
the KEEPALIVE return code in res_features.c;
There are other places that use this result code
for similar purposes at a higher level, these appear
to be left alone in 1.4, but attacked in trunk.
The reason these changes are being made in 1.4, is
that parking ends a channel's life, in some situations,
and the code in the bridge (and some other places),
was not checking the result code properly, and dereferencing
the channel pointer, which could lead to memory corruption
and crashes.
Calling the masq_park function eliminates this danger
in higher levels.
A series of previous commits have replaced some parking calls
with masq_park, but this patch puts them ALL to rest,
(except one, purposely left alone because a masquerade
is done anyway), and gets rid of the code that tests
the KEEPALIVE result, and the NOHANGUP_PEER result codes.
While bug 13820 inspired this work, this patch does
not solve all the problems mentioned there.
I have tested this patch (again) to make sure I have
not introduced regressions.
Crashes that occurred when a parked party hung up
while the parking party was listening to the numbers
of the parking stall being assigned, is eliminated.
These are the cases where parking code may be activated:
1. Feature one touch (eg. *3)
2. Feature blind xfer to parking lot (eg ##700)
3. Run Park() app from dialplan (eg sip xfer to 700)
(eg. dahdi hookflash xfer to 700)
4. Run Park via manager.
The interesting testing cases for parking are:
I. A calls B, A parks B
a. B hangs up while A is getting the numbers announced.
b. B hangs up after A gets the announcement, but
before the parking time expires
c. B waits, time expires, A is redialed,
A answers, B and A are connected, after
which, B hangs up.
d. C picks up B while still in parking lot.
II. A calls B, B parks A
a. A hangs up while B is getting the numbers announced.
b. A hangs up after B gets the announcement, but
before the parking time expires
c. A waits, time expires, B is redialed,
B answers, A and B are connected, after
which, A hangs up.
d. C picks up A while still in parking lot.
Testing this throroughly involves acting all the permutations
of I and II, in situations 1,2,3, and 4.
Since I added a few more changes (ALL references to KEEPALIVE in the bridge
code eliimated (I missed one earlier), I retested
most of the above cases, and no crashes.
H-extension weirdness.
Current h-extension execution is not completely
correct for several of the cases.
For the case where A calls B, and A parks B, the
'h' exten is run on A's channel as soon as the park
is accomplished. This is expected behavior.
But when A calls B, and B parks A, this will be
current behavior:
After B parks A, B is hung up by the system, and
the 'h' (hangup) exten gets run, but the channel
mentioned will be a derivative of A's...
Thus, if A is DAHDI/1, and B is DAHDI/2,
the h-extension will be run on channel
Parked/DAHDI/1-1<ZOMBIE>, and the
start/answer/end info will be those
relating to Channel A.
And, in the case where A is reconnected to
B after the park time expires, when both parties
hang up after the joyful reunion, no h-exten
will be run at all.
In the case where C picks up A from the
parking lot, when either A or C hang up,
the h-exten will be run for the C channel.
CDR's are a separate issue, and not addressed
here.
As to WHY this strange behavior occurs,
the answer lies in the procedure followed
to accomplish handing over the channel
to the parking manager thread. This procedure
is called masquerading. In the process,
a duplicate copy of the channel is created,
and most of the active data is given to the
new copy. The original channel gets its name
changed to XXX<ZOMBIE> and keeps the PBX
information for the sake of the original
thread (preserving its role as a call
originator, if it had this role to begin
with), while the new channel is without
this info and becomes a call target (a
"peer").
In this case, the parking lot manager
thread is handed the new (masqueraded)
channel. It will not run an h-exten
on the channel if it hangs up while
in the parking lot. The h exten will
be run on the original channel instead,
in the original thread, after the bridge
completes.
See bug 13820 for our intentions as
to how to clean up the h exten behavior.
Review: http://reviewboard.digium.com/r/29/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166665 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r166568 | mmichelson | 2008-12-23 09:16:26 -0600 (Tue, 23 Dec 2008) | 12 lines
Fix a crash resulting from a datastore with inheritance but no duplicate callback
The fix for this is to simply set the newly created datastore's data pointer
to NULL if it is inherited but has no duplicate callback.
(closes issue #14113)
Reported by: francesco_r
Patches:
14113.patch uploaded by putnopvut (license 60)
Tested by: francesco_r
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166569 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch introduces a function to do careful writes on a file stream which
will handle timeouts and partial writes. It is currently used in AMI to
address the issue that has been reported. However, there are probably a few
other places where this could be used.
(closes issue #13546)
Reported by: srt
Tested by: russell
http://reviewboard.digium.com/r/104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166282 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A crash was brought up on the bugtracker. The first run through valgrind
was full of legitimate complaints of uninitialized values in func_timeout when
setting a response timeout. These were fixed but the crash persisted.
A second run through showed the real problem. The reference counting used
for filestreams was incorrect because there were some missing increments
when a frame was read from a format module.
(closes issue #14118)
Reported by: blitzrage
Patches:
14118v2.patch uploaded by putnopvut (license 60)
Tested by: blitzrage
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166267 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This function is being added as a method to allow for
an audiohook to move to a new channel during a channel
masquerade. The most obvious use for such a facility is
for MixMonitor when a transfer is performed. Prior to
the addition of this functionality, if a channel
running MixMonitor was transferred by another party, then
the recording would stop once the transfer had completed.
By using AUDIOHOOK_INHERIT, you can make MixMonitor
continue recording the call even after the transfer
has completed.
It has also been determined that since this is seen
by most as a bug fix and is not an invasive change,
this functionality will also be backported to 1.4 and
merged into the 1.6.0 branches, even though they are
feature-frozen.
(closes issue #13538)
Reported by: mbit
Patches:
13538.patch uploaded by putnopvut (license 60)
Tested by: putnopvut
Review: http://reviewboard.digium.com/r/102/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r165796 | russell | 2008-12-18 15:39:25 -0600 (Thu, 18 Dec 2008) | 11 lines
Make ast_carefulwrite() be more careful.
This patch handles some additional cases that could result in partial writes
to the file description. This was done to address complaints about partial
writes on AMI.
(issue #13546) (more changes needed to address potential problems in 1.6)
Reported by: srt
Tested by: russell
Review: http://reviewboard.digium.com/r/99/
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This is part of an effort to completely remove AST_PBX_KEEPALIVE and other
similar return codes from the source. While this usage was perfectly safe,
there are others that are problematic. Since we know ahead of time that
we do not want to PBX to destroy the channel, the PBX API has been changed
so that information can be provided as an argument, instead, thus removing
the need for the KEEPALIVE return value.
Further changes to get rid of KEEPALIVE and related code is being done by
murf. There is a patch up for that on review 29.
Review: http://reviewboard.digium.com/r/98/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165723 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r165591 | file | 2008-12-18 13:11:42 -0400 (Thu, 18 Dec 2008) | 4 lines
Only care about a compatible codec for early bridging if we are actually bridging to another channel. If we are not we actually want to bring the audio back to us.
(closes issue #13545)
Reported by: davidw
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165599 65c4cc65-6c06-0410-ace0-fbb531ad65f3
access 'struct ast_str' members, but this must only be used inside the API implementation.
(closes issue #14098)
Reported by: eliel
Patches:
ast_str.patch uploaded by eliel (license 64)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165502 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r164881 | russell | 2008-12-16 15:38:29 -0600 (Tue, 16 Dec 2008) | 9 lines
Fix an issue where DEBUG_THREADS may erroneously report that a thread
is exiting while holding a lock.
If the last lock attempt was a trylock, and it failed, it will still be in the
list of locks so that it can be reported.
(closes issue #13219)
Reported by: pj
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r164806 | russell | 2008-12-16 14:35:25 -0600 (Tue, 16 Dec 2008) | 9 lines
Add "restart gracefully" to the AMI blacklist of CLI commands.
"module unload" was already identified as a command that can not be used
from the AMI. "restart gracefully" effectively unloads all modules, and will
run in to the same problems.
(closes issue #13894)
Reported by: kernelsensei
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164807 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: toc
Tested by: murf
OK, Well this issue has had its share of flip-flopping.
I found the following:
1. the code in question, in ext_cmp1 in pbx.c, would not
allow two extensions that vary only by any dashes contained
within them, to be defined in the same context.
2. for input dialstrings, dashes are NOT ignored.
So, skipping them when sorting patterns seemed a bit silly.
Thus, you might declare ext 891 in a context, but
if you try dialing 8-9-1, it will NOT match 891.
So, I proposed to remove the code from ext_cmp1 to
skip the spaces and dashes. Just kept us from
declaring 891 and 8-9-1 in the same context,
forcing users to generate otherwise uselessly
obfuscated dialplan code to get the same effect.
Then, I tried out 1.4, and found that:
1. you can declare 891 and 8-9-1 in the
same context!
2. You can't define 891, and have 8-9-1 match
it! Nor can you define 8-9-1, and have 891
match it!
So, it appears that my proposal simply restores
the pbx to behaving as it did in 1.4.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164801 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r164736 | russell | 2008-12-16 11:06:29 -0600 (Tue, 16 Dec 2008) | 14 lines
Fix memory leak and invalid reporting issues with DEBUG_THREADLOCALS.
One issue was that the ast_mutex_* API was being used within the context of the
thread local data destructors. We would go off and allocate more thread local data
while the pthread lib was in the middle of destroying it all. This led to a memory
leak.
Another issue was an invalid argument being provided to the the object_add
API call.
(closes issue #13678)
Reported by: ys
Tested by: Russell
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r164201 | russell | 2008-12-15 08:31:37 -0600 (Mon, 15 Dec 2008) | 31 lines
Handle a case where a call can be bridged to a channel that is still ringing.
The issue that was reported was about a case where a RINGING channel got
redirected to an extension to pick up a call from parking. Once the parked
call got taken out of parking, it heard silence until the other side answered.
Ideally, the caller that was parked would get a ringing indication. This patch
fixes this case so that the caller receives ringback once it comes out of
parking until the other side answers.
The fixes are:
- Make sure we remember that a channel was an outgoing channel when doing
a masquerade. This prevents an erroneous ast_answer() call on the channel,
which causes a bogus 200 OK to be sent in the case of SIP.
- Add some additional comments to explain related parts of code.
- Update the handling of the ast_channel visible_indication field. Storing
values that are not stateful is pointless. Control frames that are events
or commands should be ignored.
- When a bridge first starts, check to see if the peer channel needs to be
given ringing indication because the calling side is still ringing.
- Rework ast_indicate_data() a bit for the sake of readability.
(closes issue #13747)
Reported by: davidw
Tested by: russell
Review: http://reviewboard.digium.com/r/90/
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r163761 | tilghman | 2008-12-12 16:03:10 -0600 (Fri, 12 Dec 2008) | 7 lines
Simple fix for Ctrl-C not immediately exiting Asterisk, but also add a
pointer inside editline to look back to asterisk.c, so others don't spend
as much time as I did looking (in the wrong place) for the appropriate
function.
Reported by: ZX81, via the #asterisk-users channel
Fixed by: me (license 14)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163762 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The name "ser" was used in a lot of places. However, it is a relic from when
the struct was a server_instance, not a session_instance. It was renamed since
it represents both a server or client connection.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163670 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r163448 | russell | 2008-12-12 07:44:08 -0600 (Fri, 12 Dec 2008) | 26 lines
Resolve issues that could cause DTMF to be processed out of order.
These changes come from team/russell/issue_12658
1) Change autoservice to put digits on the head of the channel's frame readq
instead of the tail. If there were frames on the readq that autoservice
had not yet read, the previous code would have resulted in out of order
processing. This required a new API call to queue a frame to the head
of the queue instead of the tail.
2) Change up the processing of DTMF in ast_read(). Some of the problems
were the result of having two sources of pending DTMF frames. There
was the dtmfq and the more generic readq. Both were used for pending
DTMF in various scenarios. Simplifying things to only use the frame
readq avoids some of the problems.
3) Fix a bug where a DTMF END frame could get passed through when it
shouldn't have. If code set END_DTMF_ONLY in the middle of digit emulation,
and a digit arrived before emulation was complete, digits would get
processed out of order.
(closes issue #12658)
Reported by: dimas
Tested by: russell, file
Review: http://reviewboard.digium.com/r/85/
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r163383 | tilghman | 2008-12-11 17:35:55 -0600 (Thu, 11 Dec 2008) | 9 lines
When a Ctrl-C or Ctrl-D ends a remote console, on certain shells, the terminal
is messed up. By intercepting those events with a signal handler in the remote
console, we can avoid those issues.
(closes issue #13464)
Reported by: tzafrir
Patches:
20081110__bug13464.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage
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The conversion to use ast_check_hangup() everywhere instead of checking the softhangup
flag directly introduced this problem. The issue is that ast_check_hangup() checked
for tech_pvt to be NULL. Unfortunately, this will be NULL is some valid circumstances,
such as with a dummy channel.
The fix is simple. Don't check tech_pvt. It's pointless, because the code path that
sets this to NULL is when the channel hangup callback gets called. This happens inside
of ast_hangup(), which is the same function responsible for freeing the channel. Any
code calling ast_check_hangup() better not be calling it after that point, and if so,
we have a bigger problem at hand.
(closes issue #14035)
Reported by: erogoza
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163171 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r163092 | russell | 2008-12-11 10:54:51 -0600 (Thu, 11 Dec 2008) | 11 lines
Fix an issue that made it so you could only have a single caller executing
a custom feature at a time. This was especially problematic when custom
features ran for any appreciable amount of time.
The fix turned out to be quite simple. The dynamic features are now stored
in a read/write list instead of a list using a mutex.
(closes issue #13478)
Reported by: neutrino88
Fix suggested by file
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163094 65c4cc65-6c06-0410-ace0-fbb531ad65f3
channel variables, and in any case, was redundant;
pbx_substitute_variables_helper ALREADY does substitution for global
variables.
(closes issue #13327)
Reported by: pj
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162922 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r162653 | file | 2008-12-10 12:05:29 -0400 (Wed, 10 Dec 2008) | 6 lines
Increment the sequence number on the end packets for RFC2833. After reading the RFC some more and doing some testing I agree with this change.
(closes issue #12983)
Reported by: vt
Patches:
dtmf_inc_seqnum_on_end_pkts.diff uploaded by vt (license 520)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162656 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After the nightly update of the documentation on asterisk.org, I'll post
an update to asterisk-dev with a pointer to the changes. This covers some
release branch and commit policy information. None of this should be a
surprise, since it's just documenting what we have already been doing.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162418 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r162413 | russell | 2008-12-09 16:17:39 -0600 (Tue, 09 Dec 2008) | 8 lines
Remove the test_for_thread_safety() function completely.
The test is not valid. Besides, if we actually suspected that recursive
mutexes were not working, we would get a ton of LOG_ERROR messages when
DEBUG_THREADS is turned on.
(inspired by a discussion on the asterisk-dev list)
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r162265 | mmichelson | 2008-12-09 14:28:44 -0600 (Tue, 09 Dec 2008) | 6 lines
If we fail to start a thread for the pbx to run in, we need to
be sure to decrease the number of active calls on the system.
This fix may relate to ABE-1713, but it is not certain yet.
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r162204 | file | 2008-12-09 15:47:07 -0400 (Tue, 09 Dec 2008) | 7 lines
Make sure that the timestamp for DTMF is not the same as the previous voice frame and do not send audio when transmitting DTMF as this confuses some equipment.
(closes issue #13209)
Reported by: ip-rob
Patches:
13209.diff uploaded by file (license 11)
Tested by: ip-rob, bujones
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r161948 | russell | 2008-12-09 08:52:25 -0600 (Tue, 09 Dec 2008) | 15 lines
Fix a problem with GROUP() settings on a masquerade.
The previous code carried over group settings from the old channel to the new
one. However, it did nothing with the group settings that were already on the
new channel. This patch removes all group settings that already existed on the
new channel.
I have a more complicated version of this patch which addresses only the most
blatant problem with this, which is that a channel can end up with multiple
group settings in the same category. However, I could not think of a use case
for keeping any of the group settings from the old channel, so I went this route
for now.
(closes AST-152)
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- Avoid printing the name of an argument in the [Arguments] tag if there is no description
for that argument.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
when a save occurs to a different filename, everything goes to the alternate
filename, instead of appending to the original. This is important for the
AMI command UpdateConfig.
(closes issue #13301)
Reported by: trevo
Patches:
20081113__bug13301.diff.txt uploaded by Corydon76 (license 14)
20081113__bug13301__1.6.0.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, blitzrage
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161181 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r161013 | jpeeler | 2008-12-04 12:30:41 -0600 (Thu, 04 Dec 2008) | 9 lines
(closes issue #13835)
Reported by: matt_b
Tested by: jpeeler
This mirrors a check that was present in ast_rtp_read to also be in ast_rtp_raw_write to not schedule sending the receiver report if the remote RTCP endpoint address isn't present in the RTCP structure.
Closes AST-142.
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r160943 | mmichelson | 2008-12-04 10:44:18 -0600 (Thu, 04 Dec 2008) | 15 lines
Fix a callerid parsing issue. If someone formatted callerid like the
following: "name <number>" (including the quotation marks), then the parts
would be parsed as
name: "name
number: number
This is because the closing quotation mark was not discovered since the number
and everything after was parsed out of the string earlier. Now, there is a check
to see if the closing quote occurs after the number, so that we can know if we
should strip off the opening quote on the name.
Closes AST-158
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Based on cli_permissions.conf configuration file, we are able to permit or deny
cli commands based on some patterns and the local user and group running rasterisk.
(Sorry if I missed some of the testers).
Reviewboard: http://reviewboard.digium.com/r/11/
(closes issue #11123)
Reported by: eliel
Tested by: eliel, IgorG, Laureano, otherwiseguy, mvanbaak
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------------------------------------------------------------------------
r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov 2008) | 7 lines
update dev-mode compiler flags to match the ones used by default on Ubuntu Intrepid, so all developers will see the same warnings and errors
since this branch already had some printf format attributes, enable checking for them and tag functions that didn't have them
format attributes in a consistent way
------------------------------------------------------------------------
in addition:
move some format attributes from main/utils.c to the header files they belong in, and fix up references to the relevant functions based on new compiler warnings
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159818 65c4cc65-6c06-0410-ace0-fbb531ad65f3
because SPRINTF() use non-literal format strings (which cannot be checked), move it into its own module so the rest of func_strings can benefit from format string checking
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159774 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r159316 | murf | 2008-11-25 15:41:10 -0700 (Tue, 25 Nov 2008) | 15 lines
(closes issue #12694)
Reported by: yraber
Patches:
12694.2nd.diff uploaded by murf (license 17)
Tested by: murf, laurav
Thanks to file (Joshua Colp) for his IAX fix.
the change to cdr.c allows no-answer to percolate
up into CDR's, and feels like the right place to
locate this fix; if BUSY is done here, no-answer
should be, too.
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Tilghman Lesher
Sean Bright
Eliel C. Sardanons
Kevin P. Fleming
Mark Michelson
Matthew Nicholson
Joshua Colp
Russell Bryant
Jeff Peeler
Richard Mudgett
David Vossel
Doug Bailey
Olle Johansson
Jason Parker
Steve Murphy
Michiel van Baak
Terry Wilson
Dwayne M. Hubbard
Brandon Kruse