https://origsvn.digium.com/svn/asterisk/branches/1.2
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r56504 | russell | 2007-02-23 17:20:55 -0600 (Fri, 23 Feb 2007) | 8 lines
Fix up a couple more signal handlers to not do bad things that could cause
various undesirable results. The other day, I made Asterisk deadlock by
hitting Control-C because of a bad signal handler. Now, signal handlers
just set a flag and write to an alert pipe for the flag to be handled. Then,
there is another thread that is monitoring for these flags. If being run in
console mode, it is just the main thread. If Asterisk is in the background,
a thread is created to do it.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r56406 | russell | 2007-02-23 14:17:56 -0600 (Fri, 23 Feb 2007) | 4 lines
Don't destroy mutexes before unregistering all of the entry points from the core.
Also, fix a potential memory leak from not destroying the locks for all of the
possible call numbers (about 32k of them).
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56407 65c4cc65-6c06-0410-ace0-fbb531ad65f3
voicemail is sent via email using something like sendmail. In the patch from
bug 8033 to fix various IMAP storage problems, the line endings in the email
file were changed in the code from "\n" to "\r\n". However, this breaks
sending regular voicemail to email. So, this change conditionally sets line
endings to "\r\n" only if IMAP_STORAGE is enabled.
(issue #9128, patch by jarjarbinks, modified by me to not break IMAP storage)
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r56279 | file | 2007-02-22 18:19:25 -0500 (Thu, 22 Feb 2007) | 2 lines
Always defer Agent logoff if any channels are up until they hang up. (issue #9123 reported by arbrandes)
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This batch of changes to the SLA code does a few different things.
* I made the SLA code event driven instead of having to act in a lot of busy
loops while dialing things to wait for state changes. This makes the code
more efficient and readable at the same time.
* I have implemented a couple of new features. The first is inbound trunk
ringing timeouts. This is an option that defines how long to let an incoming
call on a trunk to ring.
* I have also implemented ring timeouts for stations. They may be specified
for the entire station, meaning it is how long to let the station ring before
giving up. You can also specify a ring timeout for a specific trunk on a
station. So, you can say that you only want a specific station to ring 5
seconds if it is line1 ringing, but otherwise, there is no timeout.
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r55750 | russell | 2007-02-20 18:19:14 -0600 (Tue, 20 Feb 2007) | 9 lines
Fix random crashes when using the MeetMe application. This patch converts list
handling to use the linked list macros and most importantly, implements
reference counting on the ast_conference objects. The reference counting was
first backported from 1.4. However, that code has some problems that caused
the reference count to never hit zero. Those problems are fixed in this patch
and will be resolved in 1.4 and trunk next, with a different patch.
(issues #7647, #9073, #9106, BE-115).
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r55588 | russell | 2007-02-20 13:49:50 -0600 (Tue, 20 Feb 2007) | 3 lines
Convert a tab to spaces so that the documentation is printed out properly
aligned.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r55005 | russell | 2007-02-16 16:48:22 -0600 (Fri, 16 Feb 2007) | 9 lines
Revert the change I did in revisions 54955, 54969, and 54970, in 1.2, 1.4,
and trunk. I decided that once a conference is created from meetme.conf,
it is acceptable behavior that the pin can not be changed until the
conference goes away. I also added a note in meetme.conf to describe this
behavior.
We still have another issue in 1.4 and trunk where some conferences with no
users don't go away. That is the real bug that needs to be addressed here.
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r54999 | file | 2007-02-16 17:13:45 -0500 (Fri, 16 Feb 2007) | 2 lines
Do not send indications through ast_indicate in chan_agent but instead go directly to the technology. This way when indications are emulated they happen on the Agent channel and do not screw up formats on the channels. (issue #8439 reported by punkgode)
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