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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r94767 | tilghman | 2007-12-24 13:36:59 -0600 (Mon, 24 Dec 2007) | 5 lines
Race: we need to wait to queue a NewChannel event until after the channel is
inserted into the channel list. The reason is because some manager users
immediately queue requests from the channel when they see that event and are
confused when Asterisk reports no such channel. (Closes issue #11632)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r93625 | mmichelson | 2007-12-18 11:02:48 -0600 (Tue, 18 Dec 2007) | 6 lines
Rework deadlock avoidance used in ast_write, since it meant that agent channels which were being monitored
had one audio file recorded and one empty audio file saved.
(closes issue #11529, reported by atis patched by me)
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- Newstate event
- Now has "CalleridNum" for numeric caller id, like Newchannel
- The event does not send "<unknown>" for unknown caller IDs just an empty field
- Newstate and Newchannel events
- these have changed headers
"State" -> ChannelStateDesc Text based channel state
-> ChannelState Numeric channel state
- The events does not send "<unknown>" for unknown caller IDs just an empty field
- Newstate event
- Now has "CalleridNum" for numeric caller id, like Newchannel
- The event does not send "<unknown>" for unknown caller IDs just an empty field
- Link and Unlink events
- The "Link" and "Unlink" bridge events in channel.c are now renamed to "Bridge"
- The link state is in the bridgestate: header as "Link" or "Unlink"
- For channel.c bridges, "Bridgetype: core" is added. This opens up for
bridge events in rtp.c and channel drivers
- The "Rename" manager event has a renamed header, to use the same
terminology for the current channel as other events
- Oldname -> Channel
(Moremanager)
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ones that will actually contain data. (props to eliel on IRC for this)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #11454)
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r90876 | qwell | 2007-12-04 11:28:08 -0600 (Tue, 04 Dec 2007) | 4 lines
If we fail to create a channel after allocating a timing fd, we need to make sure to close it.
Issue 11454, patch by eliel.
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(Closes issue #11438)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r90548 | file | 2007-12-03 14:40:56 -0400 (Mon, 03 Dec 2007) | 2 lines
Preserve the indication currently playing on a channel when a masquerade operation happens. (issue #BE-88)
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r90145 | russell | 2007-11-28 18:20:34 -0600 (Wed, 28 Nov 2007) | 5 lines
This set of changes is to make some callerID handling thread-safe.
The ast_set_callerid() function needed to lock the channel. Also, the handlers
for the CALLERID() dialplan function needed to lock the channel when reading
or writing callerid values directly on the channel structure.
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r89577 | file | 2007-11-26 11:34:38 -0400 (Mon, 26 Nov 2007) | 6 lines
If channel allocation fails because the alert pipe could not be created also free the scheduler context.
(closes issue #11355)
Reported by: eliel
Patches:
main.channel.c.patch uploaded by eliel (license 64)
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well
as change name of variables to follow the rest of the naming.
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who really need it.
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build times - tested, there is no measureable difference before and
after this commit.
In this change:
use asterisk/compat.h to include a small set of system headers:
inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h,
stdlib.h, alloca.h, stdio.h
Where available, the inclusion is conditional on HAVE_FOO_H as determined
by autoconf.
Normally, source files should not include any of the above system headers,
and instead use either "asterisk.h" or "asterisk/compat.h" which does it
better.
For the time being I have left alone second-level directories
(main/db1-ast, etc.).
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through ast_mutex primitives.
To detect all occurrences, I have renamed the lock field in struct ast_channel
so it is clear that it shouldn't be used directly.
There are some uses in res/res_features.c (see details of the diff)
that are error prone as they try and lock two channels without
caring about the order (or without explaining why it is safe).
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Reported by: outtolunc
Patch by: outtolunc
Closes issue #11248
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89184 | tilghman | 2007-11-12 11:29:17 -0600 (Mon, 12 Nov 2007) | 5 lines
Fix two cases of memory corruption caused by background threads.
Reported by: atis
Patch by: tilghman
Fixes issue #10923
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- the *_CURRENT macros no longer need the list head pointer argument
- add AST_LIST_MOVE_CURRENT to encapsulate the remove/add operation when moving entries between lists
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batch of RTP changes it is no longer required for the media stream to go through Asterisk if DTMF is going over signalling. It will simply reinvite back as needed.
(closes issue #11172)
Reported by: ibc
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r88826 | mmichelson | 2007-11-05 17:29:29 -0600 (Mon, 05 Nov 2007) | 6 lines
Reworked deadlock avoidance in __ast_read. Restored audio to
callback agents.
(closes issue #11071, reported by callguy, patched by me, tested by callguy and Ted Brown)
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r88709 | russell | 2007-11-05 14:11:04 -0600 (Mon, 05 Nov 2007) | 20 lines
Merge the last bit of changes from asterisk/team/russell/readq-1.4
The issue here is that the channel frame readq handling got broken when the
code was converted to use the linked list macros. It caused corruption of the
list head and tail pointers. So, I fixed up the usage of the linked list
macros and in passing, simplified the code. I also documented what the code
is doing, as it was a bit difficult to figure out at first.
This bug showed itself with crashes showing messed up head/tail pointers for
the readq. However, there are a couple of crashes that aren't quite as obvious,
but I think may be related. So, if your bug gets closed by this commit, but
you still have a problem, please reopen or create a new bug report.
(closes issue #10936)
(closes issue #10595)
(closes issue #10368)
(closes issue #11084)
(closes issue #10040)
(closes issue #10840)
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r88624 | russell | 2007-11-05 11:46:02 -0600 (Mon, 05 Nov 2007) | 5 lines
Fix up datastore handling in ast_do_masquerade(). The code is intended to move
any channel datastores from the old channel to the new one. However, it did
not use the linked list macros properly to accomplish the task. The existing
code would only work if there was only a single datastore on the old channel.
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details and examples are in include/asterisk/stringfields.h.
Not applicable to older branches except for 1.4 which will
receive a fix for the routines that free memory pools.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r87069 | kpfleming | 2007-10-25 18:03:11 -0500 (Thu, 25 Oct 2007) | 2 lines
appending one list to another should leave the first list empty, and not require the user to do that
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didn't make much sense
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r86750 | russell | 2007-10-22 10:52:48 -0500 (Mon, 22 Oct 2007) | 8 lines
Don't leak a frame in the case that an END frame is received and the time since
the BEGIN is less than that of the defined minimum DTMF duration.
(closes issue #11051)
Reported by: casper
Patches:
channel.c.86664.diff uploaded by casper (license 55)
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r86663 | file | 2007-10-22 11:27:03 -0300 (Mon, 22 Oct 2007) | 6 lines
Move log message to before the frame it references is freed.
(closes issue #11050)
Reported by: slavon
Patches:
channel.c.86662.diff uploaded by casper (license 55)
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Closes issue #11039, as suggested by seanbright.
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r86330 | russell | 2007-10-18 13:03:10 -0500 (Thu, 18 Oct 2007) | 10 lines
The channel needs to stay locked while running timer callbacks, as they access
and modify channel data that may change elsewhere. I went through every timer
callback in the source tree to make sure that none of them did any additional
locking that could introduce deadlocks, and all is well.
(closes issue #10765)
Reported by: Ivan
Patches:
ast_1_4_11_svn_patch_channel_rc.diff uploaded by Ivan (license 229)
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(closes issue #10724)
Reported by: eliel
Patches:
chan_skinny.c.patch uploaded by eliel (license 64)
chan_oss.c.patch uploaded by eliel (license 64)
chan_mgcp.c.patch2 uploaded by eliel (license 64)
pbx_config.c.patch uploaded by seanbright (license 71)
iax2-provision.c.patch uploaded by eliel (license 64)
chan_gtalk.c.patch uploaded by eliel (license 64)
pbx_ael.c.patch uploaded by seanbright (license 71)
file.c.patch uploaded by seanbright (license 71)
image.c.patch uploaded by seanbright (license 71)
cli.c.patch uploaded by moy (license 222)
astobj2.c.patch uploaded by moy (license 222)
asterisk.c.patch uploaded by moy (license 222)
res_limit.c.patch uploaded by seanbright (license 71)
res_convert.c.patch uploaded by seanbright (license 71)
res_crypto.c.patch uploaded by seanbright (license 71)
app_osplookup.c.patch uploaded by seanbright (license 71)
app_rpt.c.patch uploaded by seanbright (license 71)
app_mixmonitor.c.patch uploaded by seanbright (license 71)
channel.c.patch uploaded by seanbright (license 71)
translate.c.patch uploaded by seanbright (license 71)
udptl.c.patch uploaded by seanbright (license 71)
threadstorage.c.patch uploaded by seanbright (license 71)
db.c.patch uploaded by seanbright (license 71)
cdr.c.patch uploaded by moy (license 222)
pbd_dundi.c.patch uploaded by moy (license 222)
app_osplookup-rev83558.patch uploaded by moy (license 222)
res_clioriginate.c.patch uploaded by moy (license 222)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r85158 | tilghman | 2007-10-09 16:55:06 -0500 (Tue, 09 Oct 2007) | 5 lines
This commit fixes the following issues:
- Deadlock in ast_write (issue #10406)
- Deadlock in ast_read (issue #10406)
- Possible mutex initialization error in lock.h (issue #10571)
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r84990 | file | 2007-10-08 12:03:07 -0300 (Mon, 08 Oct 2007) | 4 lines
Don't keep trying to native bridge if either of the channels are involved in a masquerade operation to be done.
(closes issue #10696)
Reported by: tbelder
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r84274 | dhubbard | 2007-10-01 16:25:37 -0500 (Mon, 01 Oct 2007) | 1 line
moved get_base_channel() code from action_redirect to ast_channel_masquerade() for issue 7706 and BE-160
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r84049 | tilghman | 2007-09-28 00:30:22 -0500 (Fri, 28 Sep 2007) | 3 lines
Avoid a deadlock with ALL of the locks in the masquerade function, not just the
pairs of channels. (Closes issue #10406)
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r83432 | russell | 2007-09-21 09:37:20 -0500 (Fri, 21 Sep 2007) | 4 lines
gcc 4.2 has a new set of warnings dealing with cosnt pointers. This set of
changes gets all of Asterisk (minus chan_alsa for now) to compile with gcc 4.2.
(closes issue #10774, patch from qwell)
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free the frame we get in ast_write.
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r81437 | file | 2007-09-04 10:46:23 -0300 (Tue, 04 Sep 2007) | 4 lines
(closes issue #10476)
Reported by: mdu113
Only look for the end of a digit when waiting for a digit. This in turn disables emulation in the core.
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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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* 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: 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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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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of the DTMF digit(s) to be specified and make the SendDTMF application have the capability to use it.
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