https://origsvn.digium.com/svn/asterisk/branches/1.4
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r62797 | kpfleming | 2007-05-02 19:57:23 -0400 (Wed, 02 May 2007) | 7 lines
improve static Realtime config loading from PostgreSQL:
don't request sorting on fields that are pointless to sort on
use ast_build_string() instead of snprintf()
don't request the list of fieldnames that resulted from the query when we both knew what they were before we ran the query _AND_ we aren't going to do anything with them anyway
(patch by me, inspired by blitzrage's bug report about res_config_odbc)
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r62807 | kpfleming | 2007-05-02 20:02:57 -0400 (Wed, 02 May 2007) | 15 lines
Merged revisions 62796 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r62796 | kpfleming | 2007-05-02 19:53:46 -0400 (Wed, 02 May 2007) | 7 lines
increase reliability and efficiency of static Realtime config loading via ODBC:
don't request fields we aren't going to use
don't request sorting on fields that are pointless to sort on
explicitly request the fields we want, because we can't expect the database to always return them in the order they were created
(reported by blitzrage in person (!), patch by me)
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add some new options to control what happens when you hangup on an attended
transfer before the target extension answers the transferred channel. You
can now have it send the transferee back to the transferer.
(issue #8413, patch from sergee with very minor modifications by me)
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r62548 | russell | 2007-05-01 16:57:10 -0500 (Tue, 01 May 2007) | 12 lines
Merged revisions 62547 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r62547 | russell | 2007-05-01 16:55:19 -0500 (Tue, 01 May 2007) | 4 lines
Remove an unnecessary check that makes it so if you hang up after doing an
attended transfer before the target extension answers the channel, the transfer
is not successful. (issue #9338, patch by svanlund)
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This set of changes introduces a new generic event API for use within Asterisk.
I am still working on a way for events to be shared between servers, but this
part is ready and can already be used inside of Asterisk.
This set of changes introduces the first use of the API, as well. I have
restructured the way that MWI (message waiting indication) is handled. It is
now event based instead of polling based. For example, if there are a bunch
of SIP phones subscribed to mailboxes, then chan_sip will not have to
constantly poll the mailboxes for changes. app_voicemail will generate events
when changes occur.
See UPGRADE.txt and CHANGES for some more information on the effects of these
changes from the user perspective. For developer information, see the text in
include/asterisk/event.h.
As always, additional feedback is welcome on the asterisk-dev mailing list.
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This patch adds a "Bridge" Manager action, as well as a "Bridge" dialplan
application. The manager action will allow you to steal two active channels
in the system and bridge them together. Then, the one that did not hang up
will continue in the dialplan. Using the application will bridge the calling
channel to an arbitrary channel in the system. Whichever channel does not
hang up here will continue in the dialplan, as well.
This patch has been touched by a bunch of people over the course of a couple
years. Please forgive me if I have missed your name in the history of things.
The most recent patch came from issue #5841, but there is also a reference to
an earlier version of this patch from issue #4297. The people involved in writing
and/or reviewing the code include at least: twisted, mflorrel, heath1444, davetroy,
tim_ringenbach, moy, tmancill, serge-v, and me. There are also positive test
reports from many people.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r59363 | russell | 2007-03-29 12:43:52 -0500 (Thu, 29 Mar 2007) | 6 lines
When building a response to a subscription, the "from" must be the full Jabber
ID. This fixes some problems where jabber users are not able to add their
Asterisk account to their user list, since they are unable to get Asterisk
to approve their subscription. (issue #8210, reported by caspy, and verified
by bradtem)
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* Add new module, cdr_sqlite3_custom which allows logging custom CDRs into a
SQLite3 database. (issue #7149, alerios)
* Add new module, res_config_sqlite, which adds realtime database configuration
support for SQLite version 2. I decided that this was ok since we didn't have
any realtime support for version 3. If someone ports this to version 3, then
version 2 support can be removed or marked deprecated.
(issue #7790, rbarun_proformatique)
* Mark cdr_sqlite as deprecated in favor of cdr_sqlite3_custom.
Also, note that there were other modules on the bug tracker that did not make
the cut because they provided some duplicated functionality. Those are:
* cdr_sqlite3 (issue #6754, moy)
* cdr_sqlite3 (issue #8694, bsd)
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external libraries and URLs to these. Please help me add these
references.
We might want to create a similar macro "\linuxpackage" to list
the needed Linux packages in popular distributions.
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pretty cool things.
First, you can get the device state of anything in the dialplan:
NoOp(SIP/mypeer has state ${DEVSTATE(SIP/mypeer)})
NoOp(The conference room 1234 has state ${DEVSTATE(MeetMe:1234)})
Most importantly, this allows you to create custom device states so you can
control phone lamps directly from the dialplan.
Set(DEVSTATE(Custom:mycustomlamp)=BUSY)
...
exten => mycustomlamp,hint,Custom:mycustomlamp
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previously set are erroneously still set (Bug 6701). After discussion,
it was determined this should only be changed in trunk.
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r49742 | qwell | 2007-01-05 18:24:38 -0600 (Fri, 05 Jan 2007) | 7 lines
Save 1 whopping byte of allocated memory!
This looks like it may have been a chicken/egg scenario..
You had to call a cleanup func, because everything was allocated.
Then since you had to call a cleanup func, you were forced to allocate - ie; strdup("").
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defined in indications.h to ind_tone_zone_sound and ind_tone_zone,
to avoid conflicts with the structs with the same names
defined in tonezone.h
Hope i haven't missed any instance.
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r48375 | tilghman | 2006-12-10 18:47:21 -0600 (Sun, 10 Dec 2006) | 13 lines
Merged revisions 48374 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48374 | tilghman | 2006-12-10 18:33:59 -0600 (Sun, 10 Dec 2006) | 5 lines
When doing a fork() and exec(), two problems existed (Issue 8086):
1) Ignored signals stayed ignored after the exec().
2) Signals could possibly fire between the fork() and exec(), causing Asterisk
signal handlers within the child to execute, which caused nasty race conditions.
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r47239 | russell | 2006-11-06 20:25:10 -0500 (Mon, 06 Nov 2006) | 13 lines
Merged revisions 47238 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r47238 | russell | 2006-11-06 20:22:58 -0500 (Mon, 06 Nov 2006) | 5 lines
If random order is enabled for files mode music on hold, set a random initial
position, instead of always starting at the first file, and doing the random
operation only when switching to the next file.
(bug reported by John Lange on the asterisk-dev mailing list)
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r46778 | russell | 2006-11-01 13:26:35 -0500 (Wed, 01 Nov 2006) | 17 lines
Merged revisions 46776 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r46776 | russell | 2006-11-01 13:24:17 -0500 (Wed, 01 Nov 2006) | 9 lines
soxmix and Asterisk expect different file extensions for certain formats. This
was already handled for the wav49 format. However, it was not handled for
ulaw and alaw. I fixed this in such a way that using the alternate extensions
for ulaw and alaw will only happen if we know we're calling soxmix, and not a
custom script defined using the MONITOR_EXEC variable. The wav49 processing
was left alone so that external scripts will see no behavior change.
(issue #7550, reported by mnicholson, proposed patch by junky, committed fix
is a bit different)
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r46363 | russell | 2006-10-27 12:39:31 -0500 (Fri, 27 Oct 2006) | 5 lines
We should always be using _exit() after a fork() or vfork() instead of exit().
This is because exit() does some extra cleanup which in some implementations
of vfork(), for example, can actually modify the state of the parent process,
causing very weird bugs or crashes. (issue #7971, Nick Gavrikov)
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r44378 | kpfleming | 2006-10-04 14:47:22 -0500 (Wed, 04 Oct 2006) | 4 lines
update thread creation code a bit
reduce standard thread stack size slightly to allow the pthreads library to allocate the stack+data and not overflow a power-of-2 allocation in the kernel and waste memory/address space
add a new stack size for 'background' threads (those that don't handle PBX calls) when LOW_MEMORY is defined
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r43779 | russell | 2006-09-27 12:55:49 -0400 (Wed, 27 Sep 2006) | 50 lines
Merged revisions 43778 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43778 | russell | 2006-09-27 12:54:30 -0400 (Wed, 27 Sep 2006) | 42 lines
Fix a problem that occurred if a user entered a digit that matched a bridge
feature that was configured using multiple digits, and the digit that was
pressed timed out in the feature digit timeout period. For example, if blind
transfer is configured as '##', and a user presses just '#'. In this situation,
the call would lock up and no longer pass any frames.
(issue #7977 reported by festr, and issue #7982 reported by michaels and
valuable input provided by mneuhauser and kuj. Fixed by me, with testing help
and peer review from Joshua Colp).
There are a couple of issues involved in this fix:
1) When ast_generic_bridge determines that there has been a timeout, it returned
AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets this result, it calls
ast_generic_bridge over again with the same timestamp for the next event.
This results in an endless loop of nothing until the call is terminated.
This is resolved by simply changing ast_generic_bridge to return
AST_BRIDGE_COMPLETE when it sees a timeout.
2) I also changed ast_channel_bridge such that if in the process of calculating
the time until the next event, it knows a timeout has already occured, to
immediately return AST_BRIDGE_COMPLETE instead of attempting to bridge the
channels anyway.
3) In the process of testing the previous two changes, I ran into a problem in
res_features where ast_channel_bridge would return because it determined
that there was a timeout. However, ast_bridge_call in res_features would
then determine by its own calculation that there was still 1 ms before the
timeout really occurs. It would then proceed, and since the bridge broke
out and did *not* return a frame, it interpreted this as the call was over
and hung up the channels.
The reason for this was because ast_bridge_call in res_features and
ast_channel_bridge in channel.c were using different times for their
calculations. channel.c uses the start_time on the bridge config, which
is the time that the feature digit was recieved. However, res_features
had another time, 'start', which was set right before calling
ast_channel_bridge. 'start' will always be slightly after start_time in the
bridge config, and sometimes enough to round up to one ms.
This is fixed by making ast_bridge_call use the same time as
ast_channel_bridge for the timeout calculation.
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now reports AST_MODULE_LOAD_DECLINE when loading if config file
is not there, also fixed an error in res_config_pgsql where it
had a non static function when it should.
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- restructured build tree and makefiles to eliminate recursion problems
- support for embedded modules
- support for static builds
- simpler cross-compilation support
- simpler module/loader interface (no exported symbols)
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- instead of defining a free() wrapper in a bunch of files, define it as
ast_free() in utils.h and remove the copies from all the files.
- centralize and abstract the code used for doing thread storage. The code
lives in threadstorage.h, with one function being implemented in utils.c.
This new API includes generic thread storage as well as special functions
for handling thread local dynamic length string buffers.
- update ast_inet_ntoa() to use the new threadstorage API
- update ast_state2str() to use the new threadstorage API
- update ast_cli() to use the new threadstorage API
- Modify manager_event() to use thread storage. Instead of using a buffer of
4096 characters as the workspace for building the manager event, use a thread
local dynamic string. Now there is no length limitation on the length of the
body of a manager event.
- Significantly simplify the handling of ast_verbose() ...
- Instead of using a static char buffer and a lock to make sure only one
thread can be using ast_verbose() at a time, use a thread local dynamic
string as the workspace for preparing the verbose message. Instead of
locking around the entire function, the only locking done now is when the
message has been built and is being deliviered to the list of registered
verbose message handlers.
- This function was doing a strdup() on every message passed to it and
keeping a queue of the last 200 messages in memory. This has been
completely removed. The only place this was used was that if there were
any messages in the verbose queue when a verbose handler was registered,
all of the messages in the queue would be fed to it. So, I just made sure
that the console verbose handler and the network verbose handler (for
remote asterisk consoles) were registered before any verbose messages.
pbx_gtkconsole and pbx_kdeconsole will now lose a few verbose messages at
startup, but I didn't feel the performance hit of this message queue was
worth saving the initial verbose output for these very rarely used modules.
- I have removed the last three arguments to the verbose handlers, leaving
only the string itself because they aren't needed anymore. For example,
ast_verbose had some logic for telling the verbose handler to add
a newline if the buffer was completely full. Now that the buffer can grow
as needed, this doesn't matter anymore.
- remove unused function, ast_verbose_dmesg() which was to dispatch the
message queue
- Convert the list of verbose handlers to use the linked list macros.
- add missing newline characters to a few ast_verbose() calls
- convert the list of log channels to use the linked list macros in logger.c
- fix close_logger() to close all of the files it opened for logging
- update ast_log() to use a thread local dynamic string for its workspace
for preparing log messages instead of a buffer of size BUFSIZ (8kB on my
system) allocated on the stack. The dynamic string in this case is limited
to only growing to a maximum size of BUFSIZ.
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r38686 | kpfleming | 2006-08-01 18:07:06 -0500 (Tue, 01 Aug 2006) | 2 lines
ensure that the 'feature digit timeout' value is taken into account when deciding how long the bridge should run (this fixes a problem report where a digit press that did not invoke a feature is never passed across the bridge)
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- Remove obsolete modules from modules.conf.sample
(make install will warn if those exist on the machine)
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- update to current loader
- update to latest build system changes to ensure snmp/agent.o is built
and linked
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-----------
- Adding devicestate providers, a new architecture to add non-channel related
device state information, like parking lots, queues, meetmes, vending machines
and Windows 98 reboots (lots of blinking on those lights)
- Adding provider for parking lots, so you can subscribe to the status of a
parking lot
- Adding provider for meetme, so you can have a blinking lamp for a meetme
( Example: exten => edvina,hint,meetme:1234 )
- Adding support for directed parking - set the PARKINGEXTEN before you manually
call Park() and you will be parked on that space. If it's occupied, dialplan
execution will continue.
This work was sponsored by Voop A/S - www.voop.com
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support the new location for zaptel.h and tonezone.h
use the dependency information output by menuselect to build Makefile rules for each module for header files and libraries
combine the common rules into a top-level Makefile.rules file
remove all (now) unnecessary stuff from subdir Makefiles
change translator API so that the newpvt() callback returns an int instead of a pointer (it no longer allocates memory)
alphabetize --with-<foo> options in configure script
enhance Net-SNMP support in configure script to provide a --with-netsnmp option
fix support for --with-pq so that if pg-config is not found when --with-pq is specified, an error will be generated
add 'optional package' usage to modules now that menuselect can output it
allow res_snmp to build by default, since the new loader changes coming soon will solve the function naming problem (and users can disable it via menuselect anyway)
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sufficient amount of time. Even if they happen to be still present, the main
Makefile will spit out a huge warning telling the user that modules not
installed by that run of "make install" are present in the modules directory.
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subdirectory instead of a for loop
- remove the FORCE target from the main Makefile and add the couple places
I used it to the .PHONY target. .PHONY does the same thing and is a built-in
more efficient way of doing it.
- add a bunch more targets to .PHONY ...
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- add a copyright header to the build_tools Makefile
- remove 'depend' from the 'all' target in agi/ and utils/ since it is handled
by the main Makefile already
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since they are targets that do not have resulting files and are never listed
as prerequisites to real targets. Using .PHONY in this manner improves make
performance by never having to check for resulting files.
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r33615 | tilghman | 2006-06-12 10:27:18 -0500 (Mon, 12 Jun 2006) | 4 lines
Move set priority up, because at this point in the code, stdout is no longer
the console. If we're unable to set priority, the error goes to Asterisk as
if it were an AGI command (issue 7335).
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file is generated. This allows a fresh checkout of asterisk to be built
and installed with the standard "./configure && make && make install".
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- Instead of always allocating 64KB of memory for every MOH class, this has
been reduced to only a single pointer per class, with more memory only
allocatted when using "files" mode, as needed
- Instead of imposing a length limit on the full filename, including full
path, of 127 characters, use PATH_MAX, the maximum length that the system
can handle
- There is no longer a limit on the number of files than can be used for a
single MOH class using "files" mode
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(issue #6491, original patch by juggie, channel variable patch by corydon,
committed patch modified to change variable name and update documentation)
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is an error executing the AGI script, or the AGI script itself returns a
non-zero value, the AGISTATUS variable will now be set to FAILURE instead of
SUCCESS.
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execution, failure, or if the channel requested hangup.
- only return -1 from the application if the application requested hangup. If
there was just a failure in execution of the AGI, just set the status
variable appropriately and move on in the dialplan.
(issue #7121, original patch by Alessandro Polverini, updated patch by srt,
committed patch is heavily modified to allow still returning -1 on hangup)
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r29196 | bweschke | 2006-05-21 10:16:59 -0500 (Sun, 21 May 2006) | 3 lines
When an application that is executed via applicationmap and exits non-zero, make sure that we pass through the correct return value from the application to make sure a segfault doesn't occur by a bridge trying to continue when it should not. Also, when executing applications via applicationmap, make sure that the application is executed against the channel whose DTMF caused it to be fired off in the first place. (part 1/2 of #7090 - this is the only fix that will be applied to both 1.2 and /trunk) acunningham and blitzrage on testing...
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So, I have removed all of the uses of AST_LIST_HEAD_INIT and replaced them
with the equivalent static initializations.
- On passing, fix a memory leak in the unload_module() function of chan_agent.
The agents list mutex was never destroyed, and the elements in the agents
list were not freed.
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- instead of allocating a frame on the stack every time the music on hold
generator is called, put a frame in the mohdata structure. Also, initialize
the parts of the frame that will never change when the mohdata struct is
allocatted and only change the necessary parts in the generator function.
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- note when the data part of the tech/data pair is missing not only when using
the app version, but the exten version as well
- instead of logging syntax errors, just output them to the CLI
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@23284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There is still a lot of similarity with builtin_blindtransfer()
which should be removed by definining functions for the common
pieces of code (eg in the first part).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@21097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Please place a Doxygen todo marker before things that needs to be checked up later
by someone - it's like /*! \todo We really need to implement this in C++ */
- option_debug checking before logging to DEBUG channel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@20790 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Kevin P. Fleming
Russell Bryant
Joshua Colp
Steve Murphy
Olle Johansson
Tilghman Lesher
Jason Parker
Paul Cadach
Luigi Rizzo
Matt O'Gorman
Mark Spencer
Matthew Fredrickson
North Antara
BJ Weschke