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
........
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
........
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
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: http://svn.digium.com/svn/asterisk/trunk@191785 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@191140 f38db490-d61c-443f-a65b-d21fe96a405b
I think it would behoove us to force "make validate-docs" to be run after the
XML documentation has been generated if dev-mode is enabled.
(closes issue #14989)
Reported by: tzafrir
Patches:
app_queue_xml.diff uploaded by tzafrir (license 46)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@190991 f38db490-d61c-443f-a65b-d21fe96a405b
This would allow for one to add a caller to a specific place in the
queue instead of just placing the caller in the back every time. To help
facilitate some interesting manipulations, a new channel variable called
QUEUEPOSITION has been added. When a caller is removed from a queue, his
position in that queue is stored in the QUEUEPOSITION variable. One such
strategy an administrator can employ is to allow for the removal of a caller
from one queue followed by the insertion of the same caller into a separate
queue in the same position.
Review: http://reviewboard.digium.com/r/189
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@190626 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@190423 f38db490-d61c-443f-a65b-d21fe96a405b
As part of the pickup process the connected line information is updated.
Part of this process does a shallow copy of the target channel's connected line
information to a local structure. Once complete the structure contents are freed.
As a result any information in the target channel's connected line information
structure is no longer valid. This change will now set the contents back to a clean
state so that the freeing of the target channel's connected line information structure
when the channel is destroyed will no longer try to double free things.
(closes issue #14839)
Reported by: lmsteffan
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@190217 f38db490-d61c-443f-a65b-d21fe96a405b
This introduces the 'z' option to app_dial. With it set, a call forward
will cancel any timeout originally set for this instance of the Dial
application.
AST-207
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@188544 f38db490-d61c-443f-a65b-d21fe96a405b
This allows for the variables to be accessed if a member macro is run.
Thanks to Grigoriy Puzankin for bringing this up on the -dev list.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@188032 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@187491 f38db490-d61c-443f-a65b-d21fe96a405b
get_cid_name should not be called with a channel lock. get_cid_name calls ast_get_hint which eventually calls pbx_find_extension. pbx_find_extension starts and stops autoservice which should not be done with a channel lock, so get_cid_name should not be called with one.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187426 f38db490-d61c-443f-a65b-d21fe96a405b
(inspired by this evening's Toronto Asterisk Users Group meeting and previous dicussions amongst various community members)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187269 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@186525 f38db490-d61c-443f-a65b-d21fe96a405b
eliel