https://origsvn.digium.com/svn/asterisk/trunk
................
r207680 | kpfleming | 2009-07-21 08:28:04 -0500 (Tue, 21 Jul 2009) | 18 lines
Merged revisions 207647 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul 2009) | 12 lines
Ensure that user-provided CFLAGS and LDFLAGS are honored.
This commit changes the build system so that user-provided flags (in ASTCFLAGS
and ASTLDFLAGS) are supplied to the compiler/linker *after* all flags provided
by the build system itself, so that the user can effectively override the
build system's flags if desired. In addition, ASTCFLAGS and ASTLDFLAGS can now
be provided *either* in the environment before running 'make', or as variable
assignments on the 'make' command line. As a result, the use of COPTS and LDOPTS
is no longer necessary, so they are no longer documented, but are still supported
so as not to break existing build systems that supply them when building Asterisk.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@207684 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r205120 | russell | 2009-07-08 10:17:19 -0500 (Wed, 08 Jul 2009) | 16 lines
Move OpenSSL initialization to a single place, make library usage thread-safe.
While doing some reading about OpenSSL, I noticed a couple of things that
needed to be improved with our usage of OpenSSL.
1) We had initialization of the library done in multiple modules. This has now
been moved to a core function that gets executed during Asterisk startup.
We already link OpenSSL into the core for TCP/TLS functionality, so this
was the most logical place to do it.
2) OpenSSL is not thread-safe by default. However, making it thread safe is
very easy. We just have to provide a couple of callbacks. One callback
returns a thread ID. The other handles locking. For more information,
start with the "Is OpenSSL thread-safe?" question on the FAQ page of
openssl.org.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@205147 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r201610 | russell | 2009-06-18 10:27:10 -0500 (Thu, 18 Jun 2009) | 36 lines
Merged revisions 201600 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r201600 | russell | 2009-06-18 10:24:31 -0500 (Thu, 18 Jun 2009) | 29 lines
Fix memory corruption and leakage related reloads of non files mode MoH classes.
For Music on Hold classes that are not files mode, meaning that we are executing
an application that will feed us audio data, we use a thread to monitor the
external application and read audio from it. This thread also makes use of the
MoH class object. In the MoH class destructor, we used pthread_cancel() to ask
the thread to exit. Unfortunately, the code did not wait to ensure that the
thread actually went away. What needed to be done is a pthread_join() to ensure
that the thread fully cleans up before we proceed. By adding this one line, we
resolve two significant problems:
1) Since the thread was never joined, it never fully goes away. So, on every
reload of non-files mode MoH, an unused thread was sticking around.
2) There was a race condition here where the application monitoring thread
could still try to access the MoH class, even though the thread executing
the MoH reload has already destroyed it.
(issue #15109)
Reported by: jvandal
(issue #15123)
Reported by: axisinternet
(issue #15195)
Reported by: amorsen
(issue AST-208)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r199743 | dvossel | 2009-06-09 11:22:04 -0500 (Tue, 09 Jun 2009) | 11 lines
module load priority
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/branches/1.6.1@199745 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r198375 | seanbright | 2009-05-30 16:11:33 -0400 (Sat, 30 May 2009) | 13 lines
Properly terminate the receive buffer before sending to iksemel.
aji_io_recv takes the maximum number of bytes to read (instead of the total
buffer size), so we have to subtract 1 from our buffer size. Without this, when
we receive packets that are larger than our buffer, iksemel will choke and
things get wonky.
(closes issue #15232)
Reported by: lp0
Patches:
05302009_res_jabber.c.patch uploaded by seanbright (license 71)
Tested by: seanbright, lp0
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@198390 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r198146 | russell | 2009-05-29 15:06:59 -0500 (Fri, 29 May 2009) | 38 lines
Resolve issues with choppy sound when using res_timing_pthread.
The situation that caused this problem was when continuous mode was being
turned on and off while a rate was set for a timing interface. A very easy
way to replicate this bug was to do a Playback() from behind a Local channel.
In this scenario, a rate gets set on the channel for doing file playback.
At the same time, continuous mode gets turned on and off about every 20 ms
as frames get queued on to the PBX side channel from the other side of the
Local channel.
Essentially, this module treated continuous mode and a set rate as mutually
exclusive states for the timer to be in. When I dug deep enough, I observed
the following pattern:
1) Set timer to tick every 20 ms.
2) Wait almost 20 ms ...
3) Continuous mode gets turned on for a queued up frame
4) Continuous mode gets turned off
5) The timer goes back to its tick per 20 ms. state but starts counting
at 0 ms.
6) Goto step 2.
Sometimes, res_timing_pthread would make it 20 ms and produce a timer tick,
but not most of the time. This is what produced the choppy sound (or sometimes
no sound at all).
Now, the module treats continuous mode and a set rate as completely independent
timer modes. They can be enabled and disabled independently of each other and
things work as expected.
(closes issue #14412)
Reported by: dome
Patches:
issue14412.diff.txt uploaded by russell (license 2)
issue14412-1.6.1.0.diff.txt uploaded by russell (license 2)
Tested by: DennisD, russell
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@198147 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r196843 | russell | 2009-05-26 13:20:57 -0500 (Tue, 26 May 2009) | 16 lines
Merged revisions 196826 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r196826 | russell | 2009-05-26 13:14:36 -0500 (Tue, 26 May 2009) | 9 lines
Resolve a file handle leak.
The frames here should have always been freed. However, out of luck, there was
never any memory leaked. However, after file streams became reference counted,
this code would leak the file stream for the file being read.
(closes issue #15181)
Reported by: jkroon
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@196869 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r190663 | russell | 2009-04-27 14:08:12 -0500 (Mon, 27 Apr 2009) | 22 lines
Merged revisions 190661-190662 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r190661 | russell | 2009-04-27 14:00:54 -0500 (Mon, 27 Apr 2009) | 9 lines
Resolve a crash in res_smdi when used with chan_dahdi.
When chan_dahdi goes to get an SMDI message, it provides no search criteria.
It just grabs the next message that arrives. This code was written with the
SMDI dialplan functions in mind, since that is now the preferred method of
using SMDI. However, this broke support of it being used from chan_dahdi.
(closes AST-212)
........
r190662 | russell | 2009-04-27 14:03:59 -0500 (Mon, 27 Apr 2009) | 2 lines
Fix a typo from 190661.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@190665 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r189464 | seanbright | 2009-04-20 17:09:59 -0400 (Mon, 20 Apr 2009) | 20 lines
Merged revisions 189462 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r189462 | seanbright | 2009-04-20 16:58:39 -0400 (Mon, 20 Apr 2009) | 13 lines
Properly handle @s within hints in AEL.
AEL was not handling the case of a device hint containing an @ symbol, which
caused parking hints (e.g. hint(park:exten@context)) to error out the parser.
This patch makes AEL treat the @ the same way it treats colon and ampersand
now, meaning the characters are included in verbatim.
(closes issue #14941)
Reported by: bpgoldsb
Patches:
bug14941.patch uploaded by seanbright (license 71)
Tested by: bpgoldsb
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@189531 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r187421 | mmichelson | 2009-04-09 12:30:39 -0500 (Thu, 09 Apr 2009) | 21 lines
Fix a crash in res_musiconhold when using cached realtime moh.
The moh_register function links an mohclass and then immediately
unrefs the class since the container now has a reference. The problem
with using realtime music on hold is that the class is allocated,
registered, and started in one fell swoop. The refcounting logic
resulted in the count being off by one. The same problem did not
happen when using a static config because the allocation and registration
of an mohclass is a separate operation from starting moh. This also did
not affect non-cached realtime moh because the classes are not registered
at all.
I also have modified res_musiconhold to use the _t_ variants of the ao2_
functions so that more info can be gleaned when attempting to trace the
refcounts. I found this to be incredibly helpful for debugging this issue
and there's no good reason to remove it.
(closes issue #14661)
Reported by: sum
........
r187424 | mmichelson | 2009-04-09 12:34:39 -0500 (Thu, 09 Apr 2009) | 3 lines
Use safe macro practices even though they really aren't necessary.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@187427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r187046 | mmichelson | 2009-04-08 11:52:20 -0500 (Wed, 08 Apr 2009) | 16 lines
Merged revisions 187045 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r187045 | mmichelson | 2009-04-08 11:52:03 -0500 (Wed, 08 Apr 2009) | 10 lines
Fix a small logical error when loading moh classes.
We were unconditionally incrementing the number of mohclasses
registered. However, we should actually only increment if the
call to moh_register was successful.
While this probably has never caused problems, I noticed it
and decided to fix it anyway.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@187048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r184673 | file | 2009-03-27 12:46:46 -0300 (Fri, 27 Mar 2009) | 7 lines
Fix speech structure leak in the AGI speech recognition integration.
The AGI dialplan applications did not destroy the speech structure automatically
if it was not destroyed by the running AGI script. They will now do this.
(issue LUMENVOX-15)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@184675 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r184339 | russell | 2009-03-25 16:57:19 -0500 (Wed, 25 Mar 2009) | 35 lines
Improve performance of the ast_event cache functionality.
This code comes from svn/asterisk/team/russell/event_performance/.
Here is a summary of the changes that have been made, in order of both
invasiveness and performance impact, from smallest to largest.
1) Asterisk 1.6.1 introduces some additional logic to be able to handle
distributed device state. This functionality comes at a cost.
One relatively minor change in this patch is that the extra processing
required for distributed device state is now completely bypassed if
it's not needed.
2) One of the things that I noticed when profiling this code was that a
_lot_ of time was spent doing string comparisons. I changed the way
strings are represented in an event to include a hash value at the front.
So, before doing a string comparison, we do an integer comparison on the
hash.
3) Finally, the code that handles the event cache has been re-written.
I tried to do this in a such a way that it had minimal impact on the API.
I did have to change one API call, though - ast_event_queue_and_cache().
However, the way it works now is nicer, IMO. Each type of event that
can be cached (MWI, device state) has its own hash table and rules for
hashing and comparing objects. This by far made the biggest impact on
performance.
For additional details regarding this code and how it was tested, please see the
review request.
(closes issue #14738)
Reported by: russell
Review: http://reviewboard.digium.com/r/205/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@184342 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r183766 | mmichelson | 2009-03-23 13:58:03 -0500 (Mon, 23 Mar 2009) | 13 lines
Merged revisions 183700 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r183700 | mmichelson | 2009-03-23 12:59:28 -0500 (Mon, 23 Mar 2009) | 7 lines
Fix a memory leak in res_monitor.c
The only way that this leak would occur is if Monitor were started
using the Manager interface and no File: header were given. Discovered
while reviewing the ast_channel_ao2 review request.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@183768 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009) | 52 lines
Merged revisions 182810 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines
Fix cases where the internal poll() was not being used when it needed to be.
We have seen a number of problems caused by poll() not working properly on
Mac OSX. If you search around, you'll find a number of references to using
select() instead of poll() to work around these issues. In Asterisk, we've
had poll.c which implements poll() using select() internally. However, we
were still getting reports of problems.
vadim investigated a bit and realized that at least on his system, even
though we were compiling in poll.o, the system poll() was still being used.
So, the primary purpose of this patch is to ensure that we're using the
internal poll() when we want it to be used.
The changes are:
1) Remove logic for when internal poll should be used from the Makefile.
Instead, put it in the configure script. The logic in the configure
script is the same as it was in the Makefile. Ideally, we would have
a functionality test for the problem, but that's not actually possible,
since we would have to be able to run an application on the _target_
system to test poll() behavior.
2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT
is not defined.
3) Change uses of poll() throughout the source tree to ast_poll(). I feel
that it is good practice to give the API call a new name when we are
changing its behavior and not using the system version directly in all cases.
So, normally, ast_poll() is just redefined to poll(). On systems where
AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll().
4) Change poll() in main/poll.c to be ast_internal_poll().
It's worth noting that any code that still uses poll() directly will work fine
(if they worked fine before). So, for example, out of tree modules that are
using poll() will not stop working or anything. However, for modules to work
properly on Mac OSX, ast_poll() needs to be used.
(closes issue #13404)
Reported by: agalbraith
Tested by: russell, vadim
http://reviewboard.digium.com/r/198/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@182946 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r181661 | file | 2009-03-12 13:53:52 -0300 (Thu, 12 Mar 2009) | 19 lines
Merged revisions 181659-181660 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r181659 | file | 2009-03-12 13:50:37 -0300 (Thu, 12 Mar 2009) | 8 lines
Fix another scenario where depending on configuration the stream would not get read.
For custom commands we don't know whether the audio is coming from a stream or not
so we are going to have to read the data despite no channels.
(closes issue #14416)
Reported by: caspy
........
r181660 | file | 2009-03-12 13:52:45 -0300 (Thu, 12 Mar 2009) | 2 lines
Fix logic flaw in previous commit.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@181663 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r181656 | file | 2009-03-12 13:32:20 -0300 (Thu, 12 Mar 2009) | 17 lines
Merged revisions 181655 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r181655 | file | 2009-03-12 13:29:19 -0300 (Thu, 12 Mar 2009) | 10 lines
Fix issue with streaming MOH failing if nobody is listening.
When a music class is setup to actually provide music on hold
from a stream we need to constantly read audio from it since it
will constantly be providing audio. This is now done despite there
being no channels listening to it.
(closes issue #14416)
Reported by: caspy
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@181658 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r179937 | mmichelson | 2009-03-03 14:59:16 -0600 (Tue, 03 Mar 2009) | 20 lines
Add documentation for timing modules used in Asterisk
This document specifies the timing modules available in Asterisk beginning
with Asterisk 1.6.1. The document goes into detail about the differences
between each and gives a general overview of what timing is used for in
Asterisk. There is also a section which can be used to help customize
your setup or to troubleshoot timing issues you may have.
I also added messages to the DAHDI timing test used in res_timing_dahdi.c
that points to this new documentation if people experience problems.
Big thanks to all who contributed comments on this.
(closes issue #14490)
Reported by: mmichelson
Patches:
timing.txt uploaded by mmichelson (license 60)
Review: http://reviewboard.digium.com/r/164/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@179939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r177286 | murf | 2009-02-18 16:50:57 -0700 (Wed, 18 Feb 2009) | 39 lines
Merged revisions 177225 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r177225 | murf | 2009-02-18 15:43:14 -0700 (Wed, 18 Feb 2009) | 34 lines
This patch fixes a regression of sorts that was introduced in
rev 24425.
It basically fixes AST-190/ABE-1782.
What was wrong: the user has 6000 extensions in one context; and
then 6000 contexts, one per extension. The parser could only handle
about 4893 of the 6000 extens in the single context.
This was due to the regression I mentioned. To get rid of
shift/reduce conflicts, Luigi set up right-recursive lists
for globals, context elements, switch lists, and statements.
Right recursive lists got rid of the warnings, but instead, they
use up a tremendous amount of stack space when the lists are long.
I saw this a few years back, and resolved not to fix it until
someone complained. That day has arrived!
After the changes were made, I ran the regression test suite,
and there were no problems.
I took the test case the user provided, and added 100,000
extensions to the single context, that already had 6,000 extens
in it. (I'll see your 6, and raise you 100!) It takes a few minutes
to read it all in, check it and generate code for it, but no
problems.
So, I think I can say that fundamentally, there are no longer
any limits on the number of items you can place in contexts,
statement blocks, switches, or globals, beyond your virt mem
constraints.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@177294 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r176666 | russell | 2009-02-17 15:22:40 -0600 (Tue, 17 Feb 2009) | 16 lines
Update the timing API to have better support for multiple timing interfaces.
1) Add module use count handling so that timing modules can be unloaded.
2) Implement unload_module() functions for the timing interface modules.
3) Allow multiple timing modules to be loaded, and use the one with the
highest priority value.
4) Report which timing module is being use in the "timing test" CLI command.
(closes issue #14489)
Reported by: russell
Review: http://reviewboard.digium.com/r/162/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@176675 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r176592 | tilghman | 2009-02-17 12:49:20 -0600 (Tue, 17 Feb 2009) | 4 lines
Add assertions in the quest to track down a refcount leak.
(closes issue #14485)
Reported by: davevg
........
r176642 | tilghman | 2009-02-17 15:14:18 -0600 (Tue, 17 Feb 2009) | 8 lines
Prior to masquerade, move the group definitions to the channel performing the
masq, so that the group count lingers past the bridge.
(closes issue #14275)
Reported by: kowalma
Patches:
20090216__bug14275.diff.txt uploaded by Corydon76 (license 14)
Tested by: kowalma
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@176644 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r168746 | murf | 2009-01-15 17:34:31 -0700 (Thu, 15 Jan 2009) | 20 lines
Merged revisions 168745 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r168745 | murf | 2009-01-15 17:19:12 -0700 (Thu, 15 Jan 2009) | 14 lines
This patch fixes a problem where a goto (or jump, in this case)
fails a consistency check because it can't find a matching
extension. The problem was a missing instruction to end
the range notation in the code where it converts the pattern
into a regex and uses the regex code to determine the match.
I tested using the AEL code the user supplied, and now,
the consistency check passes.
(closes issue #14141)
Reported by: dimas
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@168748 65c4cc65-6c06-0410-ace0-fbb531ad65f3
