Joshua Colp [Thu, 2 Nov 2006 16:51:27 +0000 (16:51 +0000)]
Set the AST_RWLOCK_INIT_VALUE to the PTHREAD_RWLOCK_INIT_VALUE if it is available, that way outside stuff can determine whether to use a constructor or deconstructor for initialization instead of using the init value.
Joshua Colp [Thu, 2 Nov 2006 16:28:13 +0000 (16:28 +0000)]
I'm crazy so I will add this... pthread rwlock wrappers, along with autoconf stuff that detects the presence of the initializer and the ability to set the kind of lock (in our case we rather like writer preferred locks so writer starvation doesn't occur... but on something like Darwin we don't get that)
Russell Bryant [Thu, 2 Nov 2006 14:07:48 +0000 (14:07 +0000)]
Change the buffer used in callerid_feed() and callerid_feed_jp() to be
allocated on the stack using alloca() instead of using malloc() since
they are only used locally to these functions.
Russell Bryant [Wed, 1 Nov 2006 22:35:52 +0000 (22:35 +0000)]
Merged revisions 46845 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46845 | russell | 2006-11-01 17:32:12 -0500 (Wed, 01 Nov 2006) | 5 lines
Add a check in the configure script to determine whether ld is GNU ld or not.
This is needed because module embedding only works for gnu ld. GNU ld is now
listed as a dependency for all of the module embedding options in menuselect.
(issue #8143)
Force poll() emulation for Darwin to always be on. It's too broken to consider being used. This resolves the console issue OSX users have been seeing. I would have liked to autoconf this but I haven't been able to come up with a test case that works. Que sera.
Russell Bryant [Wed, 1 Nov 2006 18:40:13 +0000 (18:40 +0000)]
Add the ability to pass options to the Dial application when using the DUNDi
switch in the dialplan by setting the DUNDIDIALARGS channel variable.
(issue #8084, patch by bluecrow76, with small modifications and documentation
updates)
Russell Bryant [Wed, 1 Nov 2006 18:29:05 +0000 (18:29 +0000)]
Merged revisions 46778 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
It's another round of chan_iax2 fixes! Should hopefully fix the deadlock issues people have been reporting. IAXtel now has qualify turned on for 800 peers and it is handling it fine.
Russell Bryant [Tue, 31 Oct 2006 15:22:28 +0000 (15:22 +0000)]
Fix the new send text manager command. There is no way this could have worked.
- Check the channel name string length to be zero, not non-zero
- Check the message string length to be zero, not non-zero
- unlock the channel *after* calling sendtext
Olle Johansson [Tue, 31 Oct 2006 08:08:56 +0000 (08:08 +0000)]
Take two, using find_resource on Kevin's suggestion.
Might need better locking support, giving up if we can't get the lock. Right now,
using existing locking in find_resource
Russell Bryant [Tue, 31 Oct 2006 06:10:59 +0000 (06:10 +0000)]
Merged revisions 46554 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46554 | russell | 2006-10-31 00:55:07 -0500 (Tue, 31 Oct 2006) | 5 lines
Add a small tweak to the code that checks to see whether destination formats
are translatable based on the source format. If we have already determined
that there is no translation path in one direction, don't bother checking the
other direction.
Steve Murphy [Mon, 30 Oct 2006 23:11:55 +0000 (23:11 +0000)]
These changes submitted by moy via bug 6992, to add a Dial 'End' event to asterisk. I include some changes to astman to cover other events that have been added.
when unregistering a translator, don't rebuild the translation matrix unless needed
when filtering formats out of an offer, ensure we check for translation ability in both directions
Olle Johansson [Mon, 30 Oct 2006 21:48:41 +0000 (21:48 +0000)]
Adding dialplan function IFMODULE, so you can create dialplans that handle
various PBX installations and checks if a module is loaded before using
it.
example IFMODULE(chan_sip3.so)
issue #6671 in the bug tracker, finally gone. Thanks to mithraen for keeping
it updated.
Olle Johansson [Mon, 30 Oct 2006 19:56:14 +0000 (19:56 +0000)]
Change name of "contact" setting to "callback" which better reflects what it
is to the person that configures asterisk. That we use it internally in the
contact header is a totally different story.
Olle Johansson [Sun, 29 Oct 2006 20:21:33 +0000 (20:21 +0000)]
Bind RTCP to the same IP as RTP.
I currently don't see this as a bug that needs to be fixed in 1.4/1.2 too,
but feel free to backport if you see it that way. RTCP now binds to
ALL IP addresses on the host, RTP to a specific address.
Olle Johansson [Sat, 28 Oct 2006 17:25:23 +0000 (17:25 +0000)]
- Don't lock the dialoglist during the whole destruction of a single SIP dialog. Only
lock when needed - when we remove the dialog from the dialog list
If this doesn't lead to severe problems, it might help with some locking issues
in 1.4/1.2.
- Remove the term "interface" as a synonym for a SIP dialog. Sorry, Mark, but no
one understands it... ;-)
Russell Bryant [Fri, 27 Oct 2006 19:04:34 +0000 (19:04 +0000)]
Merged revisions 46370 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46370 | russell | 2006-10-27 14:03:32 -0500 (Fri, 27 Oct 2006) | 4 lines
move the copy of the default settings to the global settings back out of
process_zap, so that they aren't overwritten when process_zap is called
multiple times
BJ Weschke [Fri, 27 Oct 2006 18:59:16 +0000 (18:59 +0000)]
* Added option to run macro when a queue member is connected to a caller,
see queues.conf.sample for details.
* Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
setqueueentryvar options for each queue, see queues.conf.sample for details.
(#8216, jmls reported and submitted)
Russell Bryant [Fri, 27 Oct 2006 17:42:57 +0000 (17:42 +0000)]
Merged revisions 46363 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
Russell Bryant [Fri, 27 Oct 2006 16:47:44 +0000 (16:47 +0000)]
Add the ability to customize some of the prompts used within the voicemail
application by configuring them in voicemail.conf (issue #7415, patch by
fkasumovic, with some fixes and documentation updates by myself)
Russell Bryant [Fri, 27 Oct 2006 15:44:34 +0000 (15:44 +0000)]
Merged revisions 46358 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46358 | russell | 2006-10-27 10:32:40 -0500 (Fri, 27 Oct 2006) | 5 lines
Instead of iterating all of the options once to look for jitterbuffer options,
and then again for everything else, move the processing of jitterbuffer
options into the main loop so that there are no erroneous messages about
ignoring unknown options. (issue #8226)
fixed a bug which caused chan_misdn to try to allocate 2 times the same channel on high load, which then caused instability of mISDN. removed a useless function from isdn_lib.c
........
Russell Bryant [Thu, 26 Oct 2006 16:35:34 +0000 (16:35 +0000)]
Merged revisions 46329 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46329 | russell | 2006-10-26 11:31:05 -0500 (Thu, 26 Oct 2006) | 11 lines
- If the source has no audio or no video portion, do not call powerof() to
get the format index.
- Don't run through the audio and video loops if there is no audio or video
portion of the source
If 0 is passed to powerof, it will return -1. This value of -1 was then being
used as an array index in these loops, which caused a crash on some systems.
Other than this issue, this code works as we expected it to. If a format is
not in the source, and we have to translation path to it, it is not offered in
the list of acceptable destination formats.
(fixes issue #8231)
projects / thirdparty / asterisk.git / log
