https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #10133)
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r74374 | qwell | 2007-07-10 13:39:30 -0500 (Tue, 10 Jul 2007) | 13 lines
Merged revisions 74373 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r74373 | qwell | 2007-07-10 13:37:23 -0500 (Tue, 10 Jul 2007) | 5 lines
Use res_ndestroy on systems that have it. Otherwise, use res_nclose.
This prevents a memleak on NetBSD - and possibly others.
Issue 10133, patch by me, reported and tested by scw
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@74375 f38db490-d61c-443f-a65b-d21fe96a405b
Reported by: julien23
Patches submitted by: julien23
Add the ability to disable recording the input or output streams in res_monitor.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@74164 f38db490-d61c-443f-a65b-d21fe96a405b
If the call is answered by another phone, other phones won't display the call as "missed".
You can also add an option to the dial command so that you can have a "followme"
scenario and not count the calls as "missed" when you cancel the call.
Thanks to Ramon and Frank for feedback on this feature.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@74024 f38db490-d61c-443f-a65b-d21fe96a405b
*must* write the file to the FILE *, and not the raw fd. Otherwise, it breaks
TLS support.
Thanks to rizzo for catching this!
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@72738 f38db490-d61c-443f-a65b-d21fe96a405b
Handle transferring large files from the built-in http server. Previously, the
code attempted to malloc a block as large as the file itself. Now it uses the
sendfile() system call so that the file isn't copied into userspace at all if
it is available. Otherwise, it just uses a read/write of small chunks at a time.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@72701 f38db490-d61c-443f-a65b-d21fe96a405b
Also adds a new "reporting" permission for manager, since it can be incredibly spammy.
This permission was discussed on the -dev mailing list some months back.
Issue 8613, patch by johann8384, with some minor changes by me.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@70961 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r69702 | russell | 2007-06-18 11:35:02 -0500 (Mon, 18 Jun 2007) | 6 lines
To prevent 92138749238754 more reports of "I have unixodbc installed, but
still can't build *_odbc.so!", check for ltdl directly, instead of just listing
it as another library to include in the unixodbc check in the configure script.
This also makes ltdl show up as a dependency in menuselect so people know what
to go install. (related to issue #9989, patch by me)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@69703 f38db490-d61c-443f-a65b-d21fe96a405b
the beginning of the file. Also, add a channel variable that indicates
the location in the file where the Playback was stopped.
(closes issue #7655, patch from sharkey)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@68502 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r67993 | oej | 2007-06-07 11:00:44 +0200 (Thu, 07 Jun 2007) | 6 lines
Issue #9738 - Make sure we can unload res_jabber. Patch by phsultan - thanks!
Due to a bug in the iksemel library, this will not work if you are using GTLS
in the connection. That's being investigated. If you figure out a way to handle
that without us having to patch iksemel, let us know in the bug report. Thanks.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r67716 | russell | 2007-06-06 11:55:59 -0500 (Wed, 06 Jun 2007) | 13 lines
Merged revisions 67715 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r67715 | russell | 2007-06-06 11:40:51 -0500 (Wed, 06 Jun 2007) | 5 lines
We have some bug reports showing crashes due to a double free of a channel.
Add a sanity check to ast_channel_free() to make sure we don't go on trying
to free a channel that wasn't found in the channel list.
(issue #8850, and others...)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r67492 | russell | 2007-06-05 15:53:28 -0500 (Tue, 05 Jun 2007) | 16 lines
This bug has been hanging over my head ever since I wrote this SLA code.
Every time I tried to go debug it by adding some debug output, the behavior
would change. It turns out I wasn't crazy. I had the following piece of code:
if (remove)
AST_LIST_REMOVE_CURRENT(...);
Well, AST_LIST_REMOVE_CURRENT was not wrapped in braces, so my conditional
statement didn't do much good at all. It always ran at least all of the
macro minus the first statement, so I was seeing list entries magically
disappear when they weren't supposed to.
After many hours of debugging, I have come to this extremely irritating fix. :)
(issues #9581, #9497)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r67308 | russell | 2007-06-05 10:51:53 -0500 (Tue, 05 Jun 2007) | 5 lines
When shutting down "gracefully", go through and run the unload() callbacks for
all of the modules. "stop now" is considered a non-graceful shutdown and will
not go through this process.
(issue #9804, reported by chrisost, patch by me)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@67310 f38db490-d61c-443f-a65b-d21fe96a405b
places that cared about device states were app_queue and the hint code in pbx.c.
The changes include converting it to use the Asterisk event system, as well as
other efficiency improvements.
* app_queue: This module used to register a callback into devicestate.c to
monitor device state changes. Now, it is just a subscriber to Asterisk
events with the type, device state.
* pbx.c hints: Previously, the device state processing thread in devicestate.c
would call ast_hint_state_changed() each time the state of a device changed.
Then, that code would go looking for all the hints that monitor that device,
and call their callbacks. All of this blocked the device state processing
thread. Now, the hint code is a subscriber of Asterisk events with the
type, device state. Furthermore, when this code receives a device state
change event, it queues it up to be processed by another thread so that it
doesn't block one of the event processing threads.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@66958 f38db490-d61c-443f-a65b-d21fe96a405b
places in the code where the same block of code for creating detached threads
was replicated. (patch from bbryant)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@65968 f38db490-d61c-443f-a65b-d21fe96a405b
class in musiconhold.conf, a caller on hold may press this digit to switch
to listening to that music class.
This involved adding a new callback for generators, which allow generators
to get notified of DTMF from the channel they are running on. Then, a callback
was implemented for the music on hold generators.
(patch from bbryant)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@65505 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r65200 | murf | 2007-05-18 16:06:27 -0600 (Fri, 18 May 2007) | 9 lines
Merged revisions 65172 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1 line
This update will fix the situation that occurs as described by 9717, where when several targets are specified for a dial, if any one them reports FAIL, the whole call gets FAIL, even though others were ringing OK. I rearranged the priorities, so that a new disposition, NULL, is at the lowest level, and the disposition get init'd to NULL. Then, next up is FAIL, and next up is BUSY, then NOANSWER, then ANSWERED. All the related set routines will only do so if the disposition value to be set to is greater than what's already there. This gives the intended effect. So, if all the targets are busy, you'd get BUSY for the call disposition. If all get BUSY, but one, and that one rings is not answered, you get NOANSWER. If by some freak of nature, the NULL value doesn't get overridden, then the disp2str routine will report NOANSWER as before.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@65202 f38db490-d61c-443f-a65b-d21fe96a405b
allow you to initiate an ENUM query using ENUMQUERY, and then access the
details of all of the results using ENUMRESULT. Previously, if you wanted
to access multiple results, Asterisk would have to do a new DNS lookup every
time. (patch by bbryant)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@64480 f38db490-d61c-443f-a65b-d21fe96a405b
- Add missing option to options.h
- Add missing variables to asterisk.h
- Move manager version to manager.h include file
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@63030 f38db490-d61c-443f-a65b-d21fe96a405b
file doc/qos.tex has been updated to document the new functionality.
(issue #9540, patch submitted by IgorG)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62457 f38db490-d61c-443f-a65b-d21fe96a405b
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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62292 f38db490-d61c-443f-a65b-d21fe96a405b
qwell