within app_voicemail for directory functions. It is therefore no longer
necessary for app_directory to be linked against the ODBC libraries (and it
never was necessary for app_directory to be linked against IMAP, though it
was).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@201783 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r201423 | dvossel | 2009-06-17 14:28:12 -0500 (Wed, 17 Jun 2009) | 19 lines
StopMixMonitor race condition (not giving up file immediately)
StopMixMonitor only indicates to the MixMonitor thread to stop
writing to the file. It does not guarantee that the recording's
file handle is available to the dialplan immediately after execution.
This results in a race condition. To resolve this, the filestream
pointer is placed in a datastore on the channel. When StopMixMonitor
is called, the datastore is retrieved from the channel and the
filestream is closed immediately before returning to the dialplan.
Documentation indicating the use of StopMixMonitor to free files
has been updated as well.
(closes issue #15259)
Reported by: travisghansen
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/283/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@201445 f38db490-d61c-443f-a65b-d21fe96a405b
Some applications (notably app_fax) do not need digit detection nor FAX tone
detection while they are running, and if Asterisk is using software DSPs to provide
the detection, this consumes extra CPU cycles that could be better spent on the
actual application. This patch allows applications to query and control the state
of digit and tone detection on a channel, and modifies app_fax to disable them
while the FAX operations are occurring (and re-enable digit detection afterwards).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@201139 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@201056 f38db490-d61c-443f-a65b-d21fe96a405b
Voicemail can only use one storage module at the moment.
Because it's unclear that selecting one of the storage modules
in menuselect will disable filesystem storage we now have
a FILE_STORAGE option that conflicts with the other modules.
(closes issue #15333)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@200943 f38db490-d61c-443f-a65b-d21fe96a405b
Fix up modules in the 'apps' directory, and also correct the bad example of
enum definitions in include/asterisk/app.h, which many developers followed
(thanks for reading the documentation!). In addition, add some basic usage
examples of the 'pahole' and 'pglobal' tools to the coding guidelines.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@200656 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@200519 f38db490-d61c-443f-a65b-d21fe96a405b
Move function MEETME_INFO static documentation to the new AstXML form.
(issue #15245)
Reported by: eliel
Patches:
app_meetme_static_conversion.txt uploaded by lmadsen (license 10)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@199409 f38db490-d61c-443f-a65b-d21fe96a405b
Move function MINIVMACCOUNT and MINIVMCOUNTER statis documentation to the new
AstXML form.
(issue #15245)
Reported by: eliel
Patches:
app_minivm_static_conversion.txt uploaded by lmadsen (license 10)
(with minor changes by me)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@199376 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@198727 f38db490-d61c-443f-a65b-d21fe96a405b
Updated the MixMonitor documentation for the 'b' option so that
it is more obvious that you must not optimize away the Local
channel when using this option.
(closes issue #14829)
Reported by: licedey
Tested by: mmichelson, licedey, lmadsen
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@197828 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@197543 f38db490-d61c-443f-a65b-d21fe96a405b
The MinivmMWI application was not being unregistered on unload and we were not
able to load again the module or reload it.
(closes issue #15174)
Reported by: junky
Patches:
unregister_minivm_mwi.diff uploaded by junky (license 177)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@196377 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@196308 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@196072 f38db490-d61c-443f-a65b-d21fe96a405b
Since we may have copied connected line info into the chanlist struct prior
to placing an outbound call, we need to be sure to free the allocated data
when we hang the call up.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@195763 f38db490-d61c-443f-a65b-d21fe96a405b
The command had a for loop that was guaranteed to only execute once since
the continuation operation of the loop would set the input buffer NULL. I rewrote
the loop so that its operation was more obvious, and it would set multiple variables
correctly.
I also reduced stack space required for the function, constified the input string,
and modified the function so that it would not modify the input string while I was
at it.
(closes issue #15114)
Reported by: chris-mac
Patches:
15114.patch uploaded by mmichelson (license 60)
Tested by: chris-mac
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@195316 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@195162 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r194028 | mnicholson | 2009-05-12 17:15:45 -0500 (Tue, 12 May 2009) | 16 lines
This change modifies app_queue to properly generate CDR records in failure
situations.
This involves setting a proper cdr disposition coresponding to the given
failure condition and ensuring the proper information is stored in the cdr
record.
(closes issue #13691)
Reported by: dferrer
Tested by: mnicholson
(closes issue #13637)
Reported by: atis
Tested by: atis
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@194057 f38db490-d61c-443f-a65b-d21fe96a405b
This helps to prevent odd scenarios where a queue will claim to have
taken 0 calls, but the members appear to have taken a non-zero amount.
(closes issue #15068)
Reported by: sum
Patches:
patchreset.patch uploaded by sum (license 766)
Tested by: sum
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@193349 f38db490-d61c-443f-a65b-d21fe96a405b
The local struct ast_party_connected_line connected_caller variable
was uninitialized when the copy function was called.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@192590 f38db490-d61c-443f-a65b-d21fe96a405b
seanbright