callback. This change differs from commit 127113 in that now the
channel is not set to AST_STATE_UP until after the answer callback.
(closes issue #12924)
Reported by: snyfer
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127157 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- change ast_settimeout() to honor max rate in edge cases of file playback
(this will make some warning messages go away at the end of playing back
a file)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125332 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r125132 | kpfleming | 2008-06-25 17:21:30 -0500 (Wed, 25 Jun 2008) | 10 lines
allow tonezone to live in a different place than DAHDI/Zaptel, since dahdi-tools and dahdi-linux are now separate packages and can be installed in different places
don't include tonezone.h in dahdi_compat.h, because only a couple of modules need it
get app_rpt building again after the DAHDI changes
(closes issue #12911)
Reported by: tzafrir
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
They want (char *)NULL as sentinel.
An example is OpenBSD (confirmed on 4.3) that ships with gcc 3.3.4
This commit introduces a contstant SENTINEL which is declared as:
#define SENTINEL ((char *)NULL)
All places I could test compile on my openbsd system are converted.
Update CODING-GUIDELINES to tell about this constant.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@124127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
them, and memory does not get free'd causing strange issues with SIP.
This code was originally written by russellb in the team/group/issue_11972/ branch.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123546 65c4cc65-6c06-0410-ace0-fbb531ad65f3
a memory leak when running the astobj2 test CLI command. After
searching, it appears the leak was in the command handler itself.
Each object was allocated (recount = 1) and then linked into
a container (refounct = 2). Then at the end of the function,
the container was unreffed, causing all the objects to have
their refcount decremented by one, leaving the refcount for
all objects allocated in that function at 1. I've now added
an extra unref to the mix so that the refcount equals zero
when the container is unreffed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r123271 | russell | 2008-06-17 10:48:31 -0500 (Tue, 17 Jun 2008) | 4 lines
Fix a memory leak in astobj2 that was pointed out by seanbright. When a container
got destroyed, the underlying bucket list entry for each object that was in the
container at that time did not get free'd.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: ys
Many thanks to ys for doing the research on this problem.
I didn't think it would be best to unlock the contexts
and then relock them after the remove_extension2() call,
so I added an extra arg to remove_extension2() and set it
appropriately in each call. There were not that many.
I considered forcing the code to lock the contexts before
the call to remove_extension2(), but that would require
a slightly greater degree of changes, especially since
the find_context_locked is local to pbx.c
I did a simple sanity test to make sure the code doesn't
mess things up in general.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Convert the last part of channel.c over to use the timing API. This would
not have made a difference when using the dahdi timing module. I noticed
it when trying to use another timing source. Oops. :)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r122713 | mmichelson | 2008-06-13 16:44:53 -0500 (Fri, 13 Jun 2008) | 9 lines
Short circuit the loop in autoservice_run if there are no channels to poll.
If we continued, then the result would be calling poll() with a NULL
pollfd array. While this is fine with POSIX's poll(2) system call, those
who use Asterisk's internal poll mechanism (Darwin systems) would have
a failed assertion occur when poll is called.
(related to issue #10342)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122714 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Convert chan_iax2 to use the timing API
- Convert usage of timing in the core to use the timing API instead of
using DAHDI directly
- Make a change to the timing API to add the set_rate() function
- change the timing core to use a rwlock
- merge a timing implementation, res_timing_dahdi
Basic testing was successful using res_timing_dahdi
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122523 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r122130 | tilghman | 2008-06-12 10:11:30 -0500 (Thu, 12 Jun 2008) | 4 lines
Occasionally, the alertpipe loses its nonblocking status, so detect and correct
that situation before it causes a deadlock. (Reported and tested by ctooley
via #asterisk-dev)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r122127 | murf | 2008-06-12 08:51:44 -0600 (Thu, 12 Jun 2008) | 1 line
Arkadia tried to warn me, but the code added to ast_cdr_busy, _failed, and _noanswer was redundant. Didn't spot it until I was resolving conflicts in trunk. Ugh. Redundant code removed. It wasn't harmful. Just dumb.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r122046 | murf | 2008-06-12 07:47:34 -0600 (Thu, 12 Jun 2008) | 37 lines
(closes issue #10668)
Reported by: arkadia
Tested by: murf, arkadia
Options added to forkCDR() app and the CDR() func to
remove some roadblocks for CDR applications.
The "show application ForkCDR" output was upgraded
to more fully explain the inner workings of forkCDR.
The A option was added to forkCDR to force the
CDR system to NOT change the disposition on the
original CDR, after the fork. This involves
ast_cdr_answer, _busy, _failed, and so on.
The T option was added to forkCDR to force
obedience of the cdr LOCKED flag in the
ast_cdr_end, all the disposition changing
funcs (ast_cdr_answer, etc), and in the
ast_cdr_setvar func.
The CHANGES file was updated to explain ALL
the new options added to satisfy this bug report
(and some requests made verbally and via
email, irc, etc, over the past months/year)
The 's' option was added to the CDR() func,
to force it to skip LOCKED cdr's in the
chain.
Again, the new options should be totally transparent
to existing apps! Current behavior of CDR,
forkCDR, and the rest of the CDR system should
not change one little bit. Until you add the
new options, at least!
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit merges in the rest of the code needed to support distributed device
state. There are two main parts to this commit.
Core changes:
- The device state handling in the core has been updated to understand device
state across a cluster of Asterisk servers. Every time the state of a device
changes, it looks at all of the device states on each node, and determines the
aggregate device state. That resulting device state is what is provided to
modules in Asterisk that take actions based on the state of a device.
New module, res_ais:
- A module has been written to facilitate the communication of events between
nodes in a cluster of Asterisk servers. This module uses the SAForum AIS
(Service Availability Forum Application Interface Specification) CLM and EVT
services (Cluster Management and Event) to handle this task. This module
currently supports sharing Voicemail MWI (Message Waiting Indication) and
device state events between servers. It has been tested with openais, though
other implementations of the spec do exist.
For more information on testing distributed device state, see the following doc:
- doc/distributed_devstate.txt
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit pulls in a batch of improvements and additions to the event API.
Changes include:
- the ability to dynamically build a subscription. This is useful if you're
building a subscription based on something you receive from the network,
or from options in a configuration file.
- Add tables of event types and IE types and the corresponding string
representation for implementing text based protocols that use these
events, for showing events on the CLI, reading configuration that
references event information, among other things.
- Add a table that maps IE types and the corresponding payload type.
- an API call to get the total size of an event
- an API call to get all events from the cache that match a subscription
- a new IE payload type, raw, which I used for transporting the Entity ID in
my code for handling distributed device state.
- Code improvements to reduce code duplication
- Include the Entity ID of the server that originated the event in every event
- an additional event type, DEVICE_STATE_CHANGE, to help facilitate distributed
device state. DEVICE_STATE is a state change on one server, DEVICE_STATE_CHANGE
is the aggregate device state change across all servers.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121555 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit breaks out some logic from pbx.c into a simple API. The hint
processing code had logic for taking the state from multiple devices and
turning that into the state for a single extension. So, I broke this out
and made an API that lets you take multiple device states and determine
the aggregate device state. I needed this for some core device state changes
to support distributed device state.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121501 65c4cc65-6c06-0410-ace0-fbb531ad65f3
DUNDi uses a concept called the Entity ID for unique server identifiers. I have
pulled out the handling of EIDs and made it something available to all of Asterisk.
There is now a global Entity ID that can be used for other purposes as well, such
as code providing distributed device state, which is why I did this. The global
Entity ID is set automatically, just like it was done in DUNDi, but it can also be
set in asterisk.conf. DUNDi will now use this global EID unless one is specified
in dundi.conf.
The current EID for the system can be seen in the "core show settings" CLI command.
It is also available in the dialplan via the ENTITYID variable.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121439 65c4cc65-6c06-0410-ace0-fbb531ad65f3