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'dialcontext' is only going to be set once during the channel's lifetime, so make it a string field instead of a char array
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alignment requirements, saving 36 bytes per instance on a 64-bit platform
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- Convert chan_iax2 to use the timing API
- Convert usage of timing in the core to use the timing API instead of
using DAHDI directly
- Make a change to the timing API to add the set_rate() function
- change the timing core to use a rwlock
- merge a timing implementation, res_timing_dahdi
Basic testing was successful using res_timing_dahdi
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Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow.
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- make data member of the ast_frame struct a named union instead of a void
Recently the ast_queue_hangup function got a new parameter, the hangupcause
Feedback came in that this is no good and that instead a new function should be created.
This I did.
The hangupcause was stored in the seqno member of the ast_frame struct. This is not very
elegant, and since there's already a data member that one should be used.
Problem is, this member was a void *.
Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone
wants to store another type in there in the future.
This commit is so massive, because all ast_frame.data uses have to be
altered to ast_frame.data.data
Thanks russellb and kpfleming for the feedback.
(closes issue #12674)
Reported by: mvanbaak
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r116463 | russell | 2008-05-14 16:32:00 -0500 (Wed, 14 May 2008) | 4 lines
Add ast_assert(), which can be used to handle fatal errors. It is only compiled
in if dev-mode is enabled, and only aborts if DO_CRASH is defined.
(inspired by issue #12650)
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(closes issue #10540)
Reported by: spendergrass
Patches:
20080417__bug10540.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage
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(closes issue #11328)
Reported by: rain
Patches:
20071207__pass_cause_in_hangup_control_frame.diff.txt uploaded by Corydon76 (license 14)
brought up-to-date to trunk by me
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Reported by: oej
Tested by: jpeeler
This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option.
Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r108583 | russell | 2008-03-13 16:38:16 -0500 (Thu, 13 Mar 2008) | 11 lines
Fix another issue that was causing crashes in chanspy. This introduces a new
datastore callback, called chan_fixup(). The concept is exactly like the
fixup callback that is used in the channel technology interface. This callback
gets called when the owning channel changes due to a masquerade. Before this
was introduced, if a masquerade happened on a channel being spyed on, the
channel pointer in the datastore became invalid.
(closes issue #12187)
(reported by, and lots of testing from atis)
(props to file for the help with ideas)
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channel. This uses control frames (so they can even pass across IAX2) to negotiate T38 and provided a way of getting the current status of T38 using queryoption. This should by no means cause any issues and if it does I will take responsibility for it.
(closes issue #11873)
Reported by: dimas
Patches:
v4-t38-api.patch uploaded by dimas (license 88)
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(closes issue #11268)
Reported by: moy
Patches:
chantrace-datastored-encapsulated-rev94934.patch uploaded by moy (license 222)
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suggestion from (const char *) Kevin. Thanks!
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r100581 | russell | 2008-01-28 11:15:41 -0600 (Mon, 28 Jan 2008) | 9 lines
Make some deadlock related fixes. These bugs were discovered and reported
internally at Digium by Steve Pitts.
- Fix up chan_local to ensure that the channel lock is held before the local
pvt lock.
- Don't hold the channel lock when executing the timing function, as it can
cause a deadlock when using chan_local. This actually changes the code back
to be how it was before the change for issue #10765. But, I added some other
locking that I think will prevent the problem reported there, as well.
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as a channel variable BRIDGEPVTCALLID
This is important for call tracing in log files and CDRs, so that
the SIP callID can be traced along servers.
The CHANNEL dialplan function won't work here, since the outbound
channel is gone when we need the Call-ID.
Other channel drivers may now implement the same function :-),
but this patch only supports chan_sip.so.
Inspired by (issue #11816)
Reported by: ctooley
Patch by oej
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines
A big one...
This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop.
This is accomplished by creating a datastore on the calling channel which has a linked list of all devices
dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this
progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply
be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore
is detached from the channel and destroyed.
This change also introduces some side effects to the code which I shall enumerate here:
1. Datastore inheritance has been backported from trunk into 1.4
2. A large chunk of code has been removed from app_dial. This chunk is the section of code
which handles the call forward case after the channel has been requested but before it has
been called. This was removed because call-forwarding still works fine without it, it makes the
code less error-prone should it need changing, and it made this set of changes much less painful
to just have the forwarding handled in one place in each module.
3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore
which is attached to the channel may be created and attached in either app_dial or app_queue, so they
need a common place to find the datastore info. This approach was taken in case similar datastores are
needed in the future, there will be a common place to add them.
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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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This works in much the same way as the automonitor, except that instead of using the monitor
app, it uses the mixmonitor app. By providing an 'x' or 'X' as a dial or queue option, a DTMF
sequence may be entered (as defined in features.conf) to start the one-touch mixmonitor.
This patch also introduces some new API calls to the audiohooks code for searching for an audiohook
by type and for searching for a running audiohook by type.
Big thanks to joetester for writing the initial patch, testing it and patiently waiting for it to
be committed.
(closes issue #10185, reported and patched by xmarksthespot)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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This was mostly to note whether a channel needed to be locked or not before
calling these functions. However, I added some other things, too.
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callbacks get called.
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well
as change name of variables to follow the rest of the naming.
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- Fix typo in chan_sip
- Remove changes to caller ID structure, moving it to branch (russellb)
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so we can handle multiple formats properly. This is not carved in stone,
but a proposal to start with.
We need to add support for transliterations as well as UTF8 handling,
propably with libiconv. Murf is looking into that for the dialplan.
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In this commit:
- move the ast_register/unregister_app functions to module.h
to avoid the need to include pbx.h for the simpler apps;
- move the ast_group structure to channel.h to remove the
dependency of app.h on linkedlists.h
Note, this is a long process that I am doing in small steps.
The main difficulty is that now for each subsystem we
have a single header (e.g. channel.h) included by the subsystem
provider (usually one file, e.g. channel.c) and by its clients
(dozens of them, e.g. we have some 70+ apps and 30+ functions).
This requires the clients to include all the extra headers
required by the provider (eg. lock.h, linkedlists.h, definitions
of substructures...) even though many of the clients would be
just happy with opaque struct declarations and function prototypes.
The long term plan is to eventually rectify this structure
so that the compilation can become faster, and also APIs
are more stable.
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who really need it.
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include/asterisk/autoconfig.h. Also, move the conditional include of sys/poll.h
or asterisk/poll-compat.h into asterisk/config.h instead of the two headers it
existed in before.
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but make sure that asterisk/compiler.h is included everywhere
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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r84018 | dhubbard | 2007-09-27 18:12:25 -0500 (Thu, 27 Sep 2007) | 1 line
if an Agent is redirected, the base channel should actually be redirected. This was causing multiple issues, especially issue 7706 and BE-160
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r81599 | russell | 2007-09-05 15:53:41 -0500 (Wed, 05 Sep 2007) | 11 lines
Fix an issue that can occur when you do an attended transfer to parking. If
you complete the transfer before the announcement of the parking spot finishes,
then the channel being parked will hear the remainder of the announcement.
These changes make it so that will not happen anymore.
Basically, res_features sets a flag on the channel is playing the announcement
to so that the file streaming core knows that it needs to watch out for a
channel masquerade, and if it occurs, to abort the announcement.
(closes BE-182)
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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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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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universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
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run on peer. Had to upgrade ast_flag stuff to 64 bits to do this.
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use the \retval tag for documenting return values, fixing various warnings
when generating the documentation, and various other things.
(closes issue #10203, snuffy)
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phones.
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.
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r73985 | tilghman | 2007-07-08 23:03:20 -0500 (Sun, 08 Jul 2007) | 2 lines
Doxygen formatting fixes; fixes errors while 'make progdocs'. (Closes issue #10104)
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