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new item
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issue #10598)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81387 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Reported by: jmls
Patches:
pbx.diff uploaded by jmls (license 141)
Add REASON dialplan variable for when an originated call fails and the failed extension is executed.
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right fix, as it is related to his last change.
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didn't add it to 1.4 yet, because it's not entirely clear to me if this is a bug fix or an enhancement. A lot of files were affected by small changes like ast_variable_new getting an added arg, for the file name the var was defined in; ast_category_new gets added args of filename and lineno; ast_category and ast_variable structures now record file and lineno for each entry; a list of all #include and #execs in a config file (or any of its inclusions are now kept in the ast_config struct; at save time, each entry is put back into its proper file of origin, in order. #include and #exec directives are folded in properly. Headers indicating that the file was generated, are generated also for each included file. Some changes to main/manager.c to take care of file renaming, via the UpdateConfig command. Multiple inclusions of the same file are handled by exploding these into multiple include files, uniquely named. There's probably more, but I can't remember it right now.
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ast_event_iterator_next.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r81342 | russell | 2007-08-29 10:57:29 -0500 (Wed, 29 Aug 2007) | 3 lines
If chan_h323 is not being built, don't use g++ to do the final link of Asterisk.
(in response to a question on the asterisk-dev list)
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have multiple information elements (while there was nothing preventing it before you could not actually access any except the first one).
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issue #10398)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81277 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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* Convert some spaces to tabs in func_volume
* Add a note in channel.h making it clear that none of the datastore API calls
lock the channel they are given, so the channel should be locked before
calling the functions that take a channel argument.
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Reported by: nic_bellamy
Patches:
2006-10-03_svn_44249_voicemail_lockmode_v3.patch uploaded by nic_bellamy (license 213)
Add support for configurable file locking methods. The default is "lockfile",
which is the old behavior. There is an additional option, "flock", which is
intended for use in situations where the lockfile method will not work, such as
with SMB/CIFS mounts.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r80974 | file | 2007-08-27 10:20:31 -0300 (Mon, 27 Aug 2007) | 4 lines
(closes issue #10562)
Reported by: idkpmiller
Correct jitter value output in the CLI to be as expected.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r80820 | russell | 2007-08-24 15:24:05 -0500 (Fri, 24 Aug 2007) | 7 lines
Improve the debouncing logic in the DTMF detector to fix some reliability
issues. Previously, this code used a shift register of hits and non-hits.
However, if the start of the digit isn't clean, it is possible for the
leading edge detector to miss the digit. These changes replace the flawed
shift register logic and also does the debouncing on the trailing edge as well.
(closes issue #10535, many thanks to softins for the patch)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r80789 | murf | 2007-08-24 12:52:15 -0600 (Fri, 24 Aug 2007) | 1 line
From a complaint by jmls, I realize that the message in cdr_disposition is unnecessary. To get failure disposition, just return -1; no use having more than one case do that.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r80424 | russell | 2007-08-22 17:40:27 -0500 (Wed, 22 Aug 2007) | 10 lines
When converting this code to use the list macros, I changed it so objects are
added to the head of a bucket instead of the tail. However, while looking over
code with mmichelson, we noticed that the algorithm used in ao2_iterator_next
requires that items are added to the tail. This wouldn't have caused any huge
problem, but it wasn't correct. It meant that if an object was added to a
container while you were iterating it, and it was added to the same bucket that
the current element is in, then the new object would be returned by
ao2_iterator_next, and any other objects in the bucket would be bypassed in
the traversal.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r80362 | russell | 2007-08-22 15:21:36 -0500 (Wed, 22 Aug 2007) | 34 lines
Merge changes from team/russell/iax_refcount.
This set of changes fixes problems with the handling of iax2_user and iax2_peer
objects. It was very possible for a thread to still hold a reference to one of
these objects while a reload operation tries to delete them. The fix here is to
ensure that all references to these objects are tracked so that they can't go away
while still in use.
To accomplish this, I used the astobj2 reference counted object model. This
code has been in one of Luigi Rizzo's branches for a long time and was primarily
developed by one of his students, Marta Carbone. I wanted to go ahead and bring
this in to 1.4 because there are other problems similar to the ones fixed by these
changes, so we might as well go ahead and use the new astobj if we're going to go
through all of the work necessary to fix the problems.
As a nice side benefit of these changes, peer and user handling got more efficient.
Using astobj2 lets us not hold the container lock for peers or users nearly as long
while iterating. Also, by changing a define at the top of chan_iax2.c, the objects
will be distributed in a hash table, drastically increasing lookup speed in these
containers, which will have a very big impact on systems that have a large number of
users or peers.
The use of the hash table will be made the default in trunk. It is not the default
in 1.4 because it changes the behavior slightly. Previously, since peers and users
were stored in memory in the same order they were specified in the configuration file,
you could influence peer and user matching order based on the order they are specified
in the configuration. The hash table does not guarantee any order in the container,
so this behavior will be going away. It just means that you have to be a little
more careful ensuring that peers and users are matched explicitly and not forcing
chan_iax2 to have to guess which user is the right one based on secret, host, and
access list settings, instead of simply using the username.
If you have any questions, feel free to ask on the asterisk-dev list.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r80255 | file | 2007-08-22 13:14:38 -0300 (Wed, 22 Aug 2007) | 4 lines
(closes issue #10526)
Reported by: sinistermidget
Revert commit from issue #10355 and return timestamp skew to 640.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r80166 | murf | 2007-08-21 10:36:34 -0600 (Tue, 21 Aug 2007) | 1 line
This patch solves problem 1 in 8126; it should not slow down the alaw codec, but should prevent signal degradation via multiple trips thru the codec. Fossil estimates the twice thru this codec will prevent fax from working. 4-6 times thru would result hearable, noticeable, voice degradation.
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is actually there.
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options, activated via the menuselect stuff, which defaults to the old way. non-zero sample data added. Translate tables expressed in microseconds instead of milliseconds, with 5-digit data now instead of 3, giving 2 more digits of precision.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #10430)
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r79904 | qwell | 2007-08-17 14:12:19 -0500 (Fri, 17 Aug 2007) | 11 lines
Don't send a semicolon over the wire in sip notify messages.
Caused by fix for issue 9938.
I basically took the code that existed before 9938 was fixed, and
copied it into a new function - ast_unescape_semicolon
There should be very few places this will be needed (pbx_config
does NOT need this (see issue 9938 for details))
Issue 10430, patch by me, with help/ideas from murf (thanks murf).
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in place of a very common construct. I also used it in a number of places
in chan_sip.
if (id > -1)
ast_sched_del(sched, id);
id = ast_sched_add(sched, ...);
changes to:
ast_sched_replace(id, sched, ...);
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redistributed/repartitioned to allow code re-use both inside and outside of Asterisk. This commit introduces the utils/conf2ael program, and an external config-file reader, for both normal config files, and for extensions.conf (context, exten, prio); It provides an API for programs outside of asterisk to use to play with the dialplan and config files.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r79553 | file | 2007-08-15 11:40:23 -0300 (Wed, 15 Aug 2007) | 6 lines
(closes issue #10440)
Reported by: irroot
(closes issue #10454)
Reported by: flo_turc
Increase maximum timestamp skew to 120. 20 was apparently far too low.
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Reported by: pj
Two of the three places ast_waitfor_nandfds could branch off to did not clear outfd and exception. If the calling function did not clear these there was a chance they could get a false positive on testing to see whether they were set.
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unless the outfd is initialized to -1 before calling the nandfds func
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r79099 | murf | 2007-08-10 14:53:43 -0600 (Fri, 10 Aug 2007) | 1 line
From a user complaint on #asterisk, I have forced pbx_spool to explain what reason codes mean, when they are logged
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r79049 | murf | 2007-08-10 12:25:51 -0600 (Fri, 10 Aug 2007) | 1 line
Re bug behavior mentioned in #asterisk, made this tweak to code, to prevent hundreds of log messages from being generated
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r79044 | murf | 2007-08-10 11:43:49 -0600 (Fri, 10 Aug 2007) | 1 line
This will help debug; from a question asked on #asterisk
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The way a device state change propagates is kind of silly, in my opinion. A
device state provider calls a function that indicates that the state of a
device has changed. Then, another thread goes back and calls a callback for
the device state provider to find out what the new state is before it can go
send it off to whoever cares.
I have changed it so that you can include the state that the device has changed
to in the first function call from the device state provider. This removes the
need to have to call the callback, which locks up critical containers to go find
out what the state changed to.
This change set changes the "simple" device state providers to use the new method.
This includes parking, meetme, and SLA.
I have also mostly converted chan_agent in my branch, but still have some more
things to think through before presenting the plan for converting channel drivers
to ensure all of the right events get generated ...
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r78955 | file | 2007-08-10 11:15:53 -0300 (Fri, 10 Aug 2007) | 2 lines
Don't bother having the core pass through or emulate begin DTMF frames when in an ast_waitstream. It only cares about the end of DTMF.
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scalability and is done in such a way that we should be able to add support for other poll() replacements.
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listen and manipulate the audio going through a channel.
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(closes issue #9892)
(closes issue #10395)
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