https://origsvn.digium.com/svn/asterisk/branches/1.4
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r203785 | russell | 2009-06-26 16:16:39 -0500 (Fri, 26 Jun 2009) | 15 lines
Don't fast forward past the end of a message.
This is nice change for users of the voicemail application. If someone gets a
little carried away with fast forwarding through a message, they can easily
get to the end and accidentally exit the voicemail application by hitting the
fast forward key during the following prompt.
This adds some safety by not allowing a fast forward past the end of a message.
(closes issue #14554)
Reported by: lacoursj
Patches:
21761.patch uploaded by lacoursj (license 707)
Tested by: lacoursj
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
CEL is the new system for logging channel events. This was inspired after
facing many problems trying to represent what is possible to happen to a call
in Asterisk using CDR records. For more information on CEL, see the built in
HTML or PDF documentation generated from the files in doc/tex/.
Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
work developing this code. Also, thanks to Matt Nicholson (mnicholson) and
Sean Bright (seanbright) for their assistance in the final push to get this
code ready for Asterisk trunk.
Review: https://reviewboard.asterisk.org/r/239/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Also change the default casing of the string contants to lowercase. This really
just saves us from have to lowercase them later when displaying them.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203605 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r203375 | russell | 2009-06-25 16:02:18 -0500 (Thu, 25 Jun 2009) | 9 lines
Fix a case where CDR answer time could be before the start time involving parking.
(closes issue #13794)
Reported by: davidw
Patches:
13794.patch uploaded by murf (license 17)
13794.patch.160 uploaded by murf (license 17)
Tested by: murf, dbrooks
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203376 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Modules placed in the priority heap for loading were not properly removed from the linked list. This resulted in some modules attempting to load twice.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202410 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r201828 | tilghman | 2009-06-18 19:40:41 -0500 (Thu, 18 Jun 2009) | 6 lines
If the "h" extension fails, give it another chance in main/pbx.c.
If the "h" extension fails, give it another chance in main/pbx.c, when it
returns from the bridge code. Fixes an issue where the "h" extension may
occasionally not fire, when a Dial is executed from a Macro.
Debugged in #asterisk with user tompaw.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201829 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r201450 | mmichelson | 2009-06-17 14:59:31 -0500 (Wed, 17 Jun 2009) | 9 lines
Change the datastore traversal in ast_do_masquerade to use a safe list traversal.
It is possible for datastore fixup functions to remove the datastore from the list
and free it. In particular, the queue_transfer_fixup in app_queue does this. While
I don't yet know of this causing any crashes, it certainly could.
Found while discussing a separate issue with Brian Degenhardt.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201458 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) | 11 lines
Improve support for media paths that can generate multiple frames at once.
There are various media paths in Asterisk (codec translators and UDPTL, primarily)
that can generate more than one frame to be generated when the application calling
them expects only a single frame. This patch addresses a number of those cases,
at least the primary ones to solve the known problems. In addition it removes the
broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API
functions, and cleans up various code paths affected by these changes.
https://reviewboard.asterisk.org/r/175/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201056 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Since there was only 1 bucket, and no hash function was specified, the code
actually worked perfectly fine. However, in theory, this was invalid use of
the OBJ_POINTER flag, so remove it so the code provides a better usage example.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200805 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch provides a new implementation of the optional API support defined
in asterisk/optional_api.h; this new version provides solves compatibility
issues with the use of linker version scripts for suppressing global symbols.
In addition, there is now a functional (and tested!) implementation for Mac OS/X,
so module writers no longer need to use special tests before calling optional
API functions. All future implementations must provide these same semantics,
so that module writers can rely on them.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When allocating the channel use ao2_ref(-1) to release it, instead of calling
ast_free().
Also avoid freeing structures inside that channel (on error) if they will be
released by the channel destructor being called if the reference counter reachs
0.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200108 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In the definition of new_find_extension(), the arguments 'callerid' and
'label' were swapped. The prototype declaration and all calls to the
function are ordered 'callerid' then 'label', but the function itself
was ordered 'label' then 'callerid'.
(closes issue #15303)
Reported by: JimDickenson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199957 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Also I removed an unnecessary free of a cid_name. This will be freed properly
in the channel destructor.
Reported by mnicholson in #asterisk-dev.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199923 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the option to give a module a load priority. The value represents the order in which a module's load() function is initialized. The lower the value, the higher the priority. The value is only checked if the AST_MODFLAG_LOAD_ORDER flag is set. If the AST_MODFLAG_LOAD_ORDER flag is not set, the value will never be read and the module will be given the lowest possible priority
on load. Since some modules are reliant on a timing interface, the timing modules have been given a high load priorty.
(closes issue #15191)
Reported by: alecdavis
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/262/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199743 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009) | 14 lines
Fixes issue with hints giving unexpected results.
Hints with two or more devices that include ONHOLD gave unexpected results.
(closes issue #15057)
Reported by: p_lindheimer
Patches:
onhold_trunk.diff uploaded by dvossel (license 671)
pbx.c.1.4.patch uploaded by p (license 558)
devicestate.c.trunk.patch uploaded by p (license 671)
Tested by: p_lindheimer, dvossel
Review: https://reviewboard.asterisk.org/r/254/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199298 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun 2009) | 40 lines
Safely handle AMI connections/reload requests that occur during startup.
During asterisk startup, a lock on the list of modules is obtained by the
primary thread while each module is initialized. Issue 13778 pointed out a
problem with this approach, however. Because the AMI is loaded before other
modules, it is possible for a module reload to be issued by a connected client
(via Action: Command), causing a deadlock.
The resolution for 13778 was to move initialization of the manager to happen
after the other modules had already been lodaded. While this fixed this
particular issue, it caused a problem for users (like FreePBX) who call AMI
scripts via an #exec in a configuration file (See issue 15189).
The solution I have come up with is to defer any reload requests that come in
until after the server is fully booted. When a call comes in to
ast_module_reload (from wherever) before we are fully booted, the request is
added to a queue of pending requests. Once we are done booting up, we then
execute these deferred requests in turn.
Note that I have tried to make this a bit more intelligent in that it will not
queue up more than 1 request for the same module to be reloaded, and if a
general reload request comes in ('module reload') the queue is flushed and we
only issue a single deferred reload for the entire system.
As for how this will impact existing installations - Before 13778, a reload
issued before module initialization was completed would result in a deadlock.
After 13778, you simply couldn't connect to the manager during startup (which
causes problems with #exec-that-calls-AMI configuration files). I believe this
is a good general purpose solution that won't negatively impact existing
installations.
(closes issue #15189)
(closes issue #13778)
Reported by: p_lindheimer
Patches:
06032009_15189_deferred_reloads.diff uploaded by seanbright (license 71)
Tested by: p_lindheimer, seanbright
Review: https://reviewboard.asterisk.org/r/272/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199051 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string. After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one. I have included this api call in both channel.c's ast_request_and_dial() and feature.c's feature_request_and_dial(). App_dial and app_queue already contain call forward logic specific for their application and options.
(closes issue #13630)
Reported by: festr
Review: https://reviewboard.asterisk.org/r/271/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198856 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When connected line updates are received or generated in the middle
of an application call, it is now possible to execute a macro to
manipulate the connected line data. This way, phone numbers may be
manipulated to be more presentable to users, names may be changed
for...whatever reason, or whatever else needs to be done may be.
Review: https://reviewboard.asterisk.org/r/256
AST-165
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198727 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Instead of specifying every object file, use make's magic to generate
it.
This will generate less conflicts in team branches when a new file is
added in trunk.
(closes issue #15226)
Reported by: eliel
Patches:
makefile uploaded by eliel (license 64)
Review: http://reviewboard.asterisk.org/r/269/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198139 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May 2009) | 15 lines
Use AST_CDR_NOANSWER instead of AST_CDR_NULL as the default CDR disposition.
This change also involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is used on originated channels to distinguish: them from dialed channels.
(closes issue #12946)
Reported by: meral
Patches:
null-cdr2.diff uploaded by mnicholson (license 96)
Tested by: mnicholson, dbrooks
(closes issue #15122)
Reported by: sum
Tested by: sum
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It is possible for a stringfield manager structure (and pool) structure to be allocated
as part of a larger structure allocation (using ast_calloc_with_strinfields()); when
this is done, the stringfield pool cannot be separately freed, but users of the tructure
may not be aware (and shouldn't have to be aware) of whether the pool was embedded.
This patch modifies the behavior so that they can always call ast_string_field_free_memory()
and the function will do the right thing for both embedded and non-embedded situations.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197775 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Since the message structure is allocated using ast_calloc_with_stringfields we do not need to
free the memory used for the stringfields as it will get freed when the message structure is.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May 2009) | 21 lines
Add flags to chanspy audiohook so that audio stays in sync.
There are two flags being added to the chanspy audiohook here. One
is the pre-existing AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set,
we ensure that the read and write slinfactories on the audiohook do
not skew beyond a certain tolerance.
In addition, there is a new audiohook flag added here,
AST_AUDIOHOOK_SMALL_QUEUE. With this flag set, we do not allow for
a slinfactory to build up a substantial amount of audio before
flushing it. For this particular issue, this means that the person
spying on the call will hear the conversations in real time with very
little delay in the audio.
(closes issue #13745)
Reported by: geoffs
Patches:
13745.patch uploaded by mmichelson (license 60)
Tested by: snblitz
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There is inconsistency in how we handle manager responses that are lists of
items and, unfortunately, third parties have come to rely on ActionID being on
every event within those lists instead of just keeping track of the ActionID for
the current response. This change makes CoreShowChannels include the ActionID
with each CoreShowChannel event generated as a result of it being called.
(closes issue #15001)
Reported by: sum
Patches:
patchactionid2.patch uploaded by sum (license 766)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196945 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The CoreStartupTime and CoreReloadTime name/value pairs in the CoreStatus
response message only included the time and not the date. This patch,
inspired by the reporter's patch, adds 2 new fields - CoreStartupDate and
CoreReloadDate - which contain the date portion of these values.
(closes issue #15000)
Reported by: sum
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196907 65c4cc65-6c06-0410-ace0-fbb531ad65f3
'channel originate ... application <app>' CLI command.
(And yeah, I cleaned up some whitespace in res_clioriginate.c... big whoop,
wanna fight about it!?)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196758 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A new xml element was created to manage the AMI actions documentation,
using AstXML.
To register a manager action using XML documentation it is now possible
using ast_manager_register_xml().
The CLI command 'manager show command' can be used to show the parsed
documentation.
Example manager xml documentation:
<manager name="ami action name" language="en_US">
<synopsis>
AMI action synopsis.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(...)" /> <-- for ActionID
<parameter name="header1" required="true">
<para>Description</para>
</parameter>
...
</syntax>
<description>
<para>AMI action description</para>
</description>
<see-also>
...
</see-also>
</manager>
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes:
- CLI command handlers
- CLI command handler arguments
- AGI command handlers
- AGI command handler arguments
- Dialplan application handler arguments
- Speech engine API function arguments
In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing.
Review: https://reviewboard.asterisk.org/r/251/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
Tilghman Lesher
Russell Bryant
Mark Michelson
Joshua Colp
Sean Bright
David Vossel
Terry Wilson
Ryan Brindley
Kevin P. Fleming
Eliel C. Sardanons
David Brooks
Matthew Nicholson
Jeff Peeler
Richard Mudgett