for extracting application, function, manager, and agi documentation is the wrong
one to take. The most severe problem is that the output depends on which modules
are loaded as well as compile time options, which both determine which parts are
available.
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r72933 | murf | 2007-07-02 14:16:31 -0600 (Mon, 02 Jul 2007) | 1 line
support for floating point numbers added to ast_expr2 $\[...\] exprs. Fixes bug 9508, where the expr code fails with fp numbers. The MATH function returns fp numbers by default, so this fix is considered necessary.
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*must* write the file to the FILE *, and not the raw fd. Otherwise, it breaks
TLS support.
Thanks to rizzo for catching this!
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Use weaker error checking because we have some automatically generated
files. However just mask out -Werror, because other warnings below:
-Wundef -Wstrict-prototypes -Wmissing-declarations
-Wmissing-prototypes
may actually be important and spot out real bugs.
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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.
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check dependencies for libraries.
Because these targets (e.g. minimime/libmmime.a) are real ones,
declaring them .PHONY would cause them to be rebuilt every time
(see e.g. SVN 64355).
As a workaround I am using the following CHECK_SUBDIR target:
CHECK_SUBDIR: # do nothing, just make sure that we recurse in the subdir/
minimime/libmmime.a: CHECK_SUBDIR
@cd minimime && $(MAKE) libmmime.a
which seems to do a better job than .PHONY (probably because
.PHONY forces the rebuild even if the recursive make does not think
it is necessary).
If this turns out to be the correct approach, we can then
merge it back into 1.4
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r72383 | bbryant | 2007-06-27 18:29:14 -0500 (Wed, 27 Jun 2007) | 11 lines
Merged revisions 72373 via svnmerge from
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r72373 | bbryant | 2007-06-27 18:22:13 -0500 (Wed, 27 Jun 2007) | 3 lines
Reinstating patch. This actually fixes the problem, however I was running a development branch without it and mistakenly thought it wasn't fixed.
Fixes issue #10010, and #9654: 100% CPU usage caused by an asterisk console losing it's controlling terminal.
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r72257 | file | 2007-06-27 16:25:24 -0400 (Wed, 27 Jun 2007) | 10 lines
Merged revisions 72256 via svnmerge from
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r72256 | file | 2007-06-27 16:23:24 -0400 (Wed, 27 Jun 2007) | 2 lines
I may possibly get shot for doing this... but... defer CDR processing until after the channel has been dealt with. This should eliminate all of the issues with channels going funky (SIP/PRI) when you are posting CDRs to a database that is either slow or unavailable and do not want to enable batching.
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r71063 | murf | 2007-06-22 08:10:24 -0600 (Fri, 22 Jun 2007) | 1 line
My conditions for merging amaflags info was naive; DOCUMENTATION is the default, although null is possible; theft of user-settable fields is not good. Just copy them, leave them alone.
This is for bug 10016. (plus a small fix to rtp, to elim a compiler warning (dev mode))
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r71066 | bbryant | 2007-06-22 09:53:08 -0500 (Fri, 22 Jun 2007) | 18 lines
Merged revisions 71064 via svnmerge from
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r71064 | bbryant | 2007-06-22 09:39:34 -0500 (Fri, 22 Jun 2007) | 10 lines
Fixed infinite loop when controlling terminal was lost
and return value of input function wasn't checked for
errors. This would cause 100% cpu to be taken up.
(closes issue #9654, issue #10010)
Reported by: mnicholson, and eserra
Idea for the patch from mnicholson, patched by me
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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.
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r70949 | murf | 2007-06-21 16:34:41 -0600 (Thu, 21 Jun 2007) | 9 lines
Merged revisions 70948 via svnmerge from
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r70948 | murf | 2007-06-21 16:29:50 -0600 (Thu, 21 Jun 2007) | 1 line
This little fix is in response to bug 10016, but may not cure it. The code is wrong, clearly. In a situation where you set the CDR's amaflags, and then ForkCDR, and then set the new CDR's amaflags to some other value, you will see that all CDRs have had their amaflags changed. This is not good. So I fixed it.
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r70062 | murf | 2007-06-19 12:23:23 -0600 (Tue, 19 Jun 2007) | 9 lines
Merged revisions 70053 via svnmerge from
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r70053 | murf | 2007-06-19 12:07:59 -0600 (Tue, 19 Jun 2007) | 1 line
This fixes 9246, where channel variables are not available in the 'h' exten, on a 'ZOMBIE' channel. The fix is to consolidate the channel variables during a masquerade, and then copy the merged variables back onto the clone, so the zombie has the same vars that the 'original' has.
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r69470 | qwell | 2007-06-14 18:22:51 -0500 (Thu, 14 Jun 2007) | 12 lines
Merged revisions 69469 via svnmerge from
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r69469 | qwell | 2007-06-14 18:21:45 -0500 (Thu, 14 Jun 2007) | 4 lines
Fix an issue where the line number in an unterminated comment block error message would show the wrong line number.
"Reported" to me on #asterisk (somebody posted an error message, and I happened to catch it)
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r69010 | russell | 2007-06-12 14:13:41 -0500 (Tue, 12 Jun 2007) | 12 lines
In ast_channel_make_compatible(), just return if the channels' read and write
formats already match up. There are code paths that call this function on a
pair of channels multiple times. This made calls fail that were using g729
in some cases. The reason is that codec_g729a will unregister itself from the
list of available translators will all licenses are in use. So, the first
time the function got called, the right translation path was allocated.
However, the second time it got called, the code would not find a translation
path to/from g729 and make the call fail, even if the channel actually already
had a g729 translation path allocated.
(SPD-32)
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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)
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r68354 | russell | 2007-06-07 18:14:45 -0500 (Thu, 07 Jun 2007) | 11 lines
Merged revisions 68351 via svnmerge from
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r68351 | russell | 2007-06-07 18:13:33 -0500 (Thu, 07 Jun 2007) | 3 lines
Fix a problem where saying a character wouldn't properly break out when the caller pressed '#'
(issue #8113, reported by patbaker82, patch from jamesgolovich (hey, long time no see!) and patbaker82)
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ast_log and other asterisk api functions available - I could not compile on OS/X without reverting
this.
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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
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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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old value instead of the old debug value, leading to an erroneous status message
after setting. This was purely a cosmetic issue and had no other underlying problems.
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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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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.
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places in the code where the same block of code for creating detached threads
was replicated. (patch from bbryant)
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r65842 | russell | 2007-05-24 09:49:05 -0500 (Thu, 24 May 2007) | 5 lines
Fix the calculation of the RTT for RTCP. The previous code would result in
oscillating and incorrect data. Additionally, the RTT would sometimes report
negative values due to incorrect calculations.
(issue #9601, patch from davetroy)
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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)
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r65201 | murf | 2007-05-18 16:26:51 -0600 (Fri, 18 May 2007) | 1 line
Ugh. The svnmerge didn't catch the shift from cdr.c to main/cdr.c, and neither did I. This is the remainder of the 9717 patch, the fix for the run-away FAIL status for a call
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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)
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This caused a problem with the buildinfo.o file not being able to find asterisk/build.h
This was affecting DESTDIR, but I *think* that if asterisk had never been installed before, it would've failed also.
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r63982 | qwell | 2007-05-11 15:16:17 -0500 (Fri, 11 May 2007) | 7 lines
Hide manager password from "manager show user foo".
I realize that there are other ways to get this,
but we really don't need to just show it in plain text so easily.
Issue 9273, patch by junky
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When enabled, it will set the systemname to be the hostname of the system
Issue 9713, patch by Juggie - slightly modified by me, to "failover" to localhost
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r63886 | russell | 2007-05-11 11:05:43 -0500 (Fri, 11 May 2007) | 6 lines
When MD5 authentication is not possible because there is no challenge present,
either because the Challenge action was never issued, or some other reason,
give a proper error message and return an error instead of claiming that the
user wasn't found.
(reported by jsmith on IRC)
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r63612 | russell | 2007-05-09 11:55:27 -0500 (Wed, 09 May 2007) | 5 lines
Modify ast_senddigit_begin() to use the same assumptions used elsewhere in the
code in that if a channel does not have a send_digit_begin() callback, it only
cares about DTMF END events. (pointed out by Michael Neuhauser on the
asterisk-dev list)
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r63608 | russell | 2007-05-09 11:43:50 -0500 (Wed, 09 May 2007) | 5 lines
Only call ast_senddigit_begin() in ast_senddigit() if the channel has a
send_digit_begin() callback. Checking the END_DTMF_ONLY flag was the
wrong thing to do, because that flag indicates that a *bridged* channel
only wants DTMF END events coming from this channel.
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- Add missing option to options.h
- Add missing variables to asterisk.h
- Move manager version to manager.h include file
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r62986 | kpfleming | 2007-05-03 11:38:56 -0500 (Thu, 03 May 2007) | 2 lines
improve loader a bit, by avoiding trying to initialize embedded modules twice and avoiding trying to load modules from disk when they have been loaded already during the 'preload' pass (reported by blitzrage on IRC, patch by me)
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r62942 | russell | 2007-05-03 10:23:13 -0500 (Thu, 03 May 2007) | 17 lines
Fix YADB (Yet Another DTMF Bug) ((C) Russell Bryant, 2007, TM, Patent Pending).
This set of changes came from a debugging session I had with Dwayne Hubbard.
When he called into his home FXO, ran the Echo application, and pressed a
digit, the digit would be echoed back and would never end. This is fixed,
along with a couple other little improvements.
* When chan_zap is in the middle of playing a digit to a channel, it feeds
back null frames, not voice frames. So, I have modified ast_read to check
the timing on emulated DTMF when it receives null frames, in addition to
where it was doing this on voice frames.
* Make a tweak to setting the duration on emulated DTMF digits. If there was
no duration specified, it set it to be the minimum, instead of the default.
* Instead of timing the emulated digits off of the number of samples in audio
frames that pass through, just use time values. Now there is no code in this
section that assumes 8kHz audio.
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r62789 | russell | 2007-05-02 17:59:09 -0500 (Wed, 02 May 2007) | 20 lines
Merge changes from team/russell/inband_dtmf ...
Fix some issues related to generating inband DTMF. There are two changes here:
1) The list of DTMF tones in the senddigit_begin() function explicitly
specified 100ms of the tone followed by 100ms of silence. This really
broke things with the way that Asterisk now wants complete control
over when the digit begins and ends. So, regardless of what Asterisk
really wanted to do, this was going to play out the tone at the length it
wanted to. This caused various problems like DTMF translation to inband to
be extremely unreliable.
The list of tones has been changed so that the correct DTMF tone is played
indefinitely until Asterisk tells it to stop.
2) ast_write() had to be modified to let a DTMF_END frame get processed even
when a generator is present. This is how the tone will finally get stopped.
(issues #8944, #9250, #9348, maybe others. Thanks to mdu113 from #8944 for
the testing and feedback!)
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r62689 | murf | 2007-05-02 11:10:50 -0600 (Wed, 02 May 2007) | 1 line
a)In chan_zap, set the clid, src fields in channel_alloc call. b)in the channel_alloc func, set the cid_num and name fields from the arglist[blush]. c) don't update the channel app & app data fields if you are in the 'h' extension. d)the load_module func in cdr_radius needs to return DECLINE, SUCCESS.
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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.
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r62171 | russell | 2007-04-27 11:14:11 -0500 (Fri, 27 Apr 2007) | 6 lines
If no variables were passed into pbx_substitute_variables_helper_full(), then
don't even bother creating a temporary bogus channel, since that is only for
allowing certain functions to operate on the variables as if they were on a
channel. Most importantly, this fixes a crash.
(issue #9613, reported by callguy, fixed by me)
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r61781 | russell | 2007-04-24 14:00:06 -0500 (Tue, 24 Apr 2007) | 6 lines
Improve DTMF handling in ast_read() even more in response to a discussion on
the asterisk-dev mailing list. I changed the enforced minimum length of a
digit from 100ms to 80ms. Furthermore, I made it now enforce a gap of 45ms in
between digits. These values are not configurable in a configuration file
right now, but they can be easily changed near the top of main/channel.c.
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r61774 | russell | 2007-04-24 11:16:41 -0500 (Tue, 24 Apr 2007) | 5 lines
Add a few more state changes in handle_frame_ownerless() so that the SLA code
will get notified of these changes even when an owner channel is not provided.
This isn't from a specific bug report, it's just something I noticed while
poking around.
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r61765 | russell | 2007-04-23 13:17:00 -0500 (Mon, 23 Apr 2007) | 5 lines
Some dialplan functions, such as CUT(), expect to operate on variables on a
channel. So, this little hack lets them work in places where a channel doesn't
exist, such as within DUNDi configuration.
(issue #9465, reported and patched by Corydon76, testing by blitzrage)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61766 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r61707 | qwell | 2007-04-20 16:35:27 -0500 (Fri, 20 Apr 2007) | 8 lines
Avoid invalid seqno cycling detection.
Per comment from Dave Troy:
This adds back in some simple typecasting I had in an earlier version
which I realize now may be breaking things.
Issue #9554.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61708 65c4cc65-6c06-0410-ace0-fbb531ad65f3
I started this for use with SLA but ended up deciding not to use it. However,
there is no reason not to put this part in, anyway.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Kevin P. Fleming
Russell Bryant
Steve Murphy
Joshua Colp
Luigi Rizzo
Tilghman Lesher
Brett Bryant
Jason Parker
Dwayne M. Hubbard
Olle Johansson
Mark Michelson
Doug Bailey
Claude Patry