Age | Commit message (Collapse) | Author |
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89296 | russell | 2007-11-15 11:19:28 -0600 (Thu, 15 Nov 2007) | 8 lines
Update the SLAStation application to account for the case where the SLA thread
has a call out to the station, but the user has pressed a line button to answer
the call instead of picking up the handset. If they do, the phone sends out a
new INVITE. So, the SLAStation app must check to see if it is picking up a
ringing trunk, and ensure that the other stations stop ringing.
(reported internally, patched by me, tested by mogorman)
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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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This extends the concise capabilities of this CLI command to include
listing all conferences, instead of an addition to the other sub commands
for the "meetme" command.
(closes issue #11078)
Reported by: jthomas
Patches:
meetme-concise.patch uploaded by jthomas (license 293)
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much identical to the S() and L() options to Dial(). They let you set
timeouts for the conference, as well as have warning sounds played to
let the caller know how much time is left, and when it is running out.
(closes issue #8030)
Reported by: areski
Patches:
meetme_timeout_timelimit_v2.patch uploaded by areski (license 29)
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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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r88026 | file | 2007-11-01 10:27:37 -0300 (Thu, 01 Nov 2007) | 2 lines
Fix up commit for my Zap channel with spies in Meetme fix. (thanks Tony Mountifield!)
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Reported by: eliel
Patch by: eliel
Closes issue #11135
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r87970 | file | 2007-10-31 22:53:55 -0300 (Wed, 31 Oct 2007) | 4 lines
If a Zap channel contains a spy or a spy is added take it out of the conference in kernel space and make it go through Asterisk so the spy gets audio from both sides.
(closes issue #10060)
Reported by: mparker
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(closes issue #10911, reported by junky, patched by me)
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menu will adjust this status if a user is muted. The talk request status will
be reflected in the CLI commands as well as the manager interface.
(closes issue #9418)
Reported by: imesper
Patches:
app_meetme_v2.patch uploaded by imesper (license 275)
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didn't make much sense
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Closes issue #11039, as suggested by seanbright.
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a patch for it. It replaces a bunch of simple calls to snprintf with ast_copy_string
(closes issue #10843)
Reported by: Corydon76
Patches:
2007092900_10843.diff uploaded by mvanbaak (license 7)
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have Command privilege. (closes issue #10736)
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(issue #10724)
Patches:
chan_zap.c.patch uploaded by moy (license 222)
app_queue.c.patch uploaded by eliel (license 64)
app_voicemail.c.patch uploaded by eliel (license 64)
app_meetme.c.patch uploaded by eliel (license 64)
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free to be ast_free, astmm said all calls to free were coming from utils.h
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r82286 | dhubbard | 2007-09-12 15:24:24 -0500 (Wed, 12 Sep 2007) | 1 line
remove a race condition for the creation of recordthread's, and fix a small memory leak. This closes issue# 10636
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Reported by: ruffle
Patches:
rb uploaded by ruffle (license 201)
Show whether the conference is locked or not on the CLI.
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Reported by: junky
Patches:
meet.diff uploaded by junky (license 177)
Change NOTICE message to DEBUG.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r81776 | file | 2007-09-06 16:40:37 -0300 (Thu, 06 Sep 2007) | 7 lines
(closes issue #10122)
Reported by: stevefeinstein
Patches:
meetme-unmute-manager.diff uploaded by qwell (license 4)
Tested by: stevefeinstein
After looking over the code I agree with Qwell. Setting the file descriptor to conference each time just causes a fight back and forth.
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(closes issue #10591, reported and patched by junky, with small modification by me)
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r78717 | russell | 2007-08-09 11:12:57 -0500 (Thu, 09 Aug 2007) | 7 lines
Fix a problem with the combination of the 'F' option to pass DTMF through a
conference and options that use DTMF to activate various features. The problem
was that the BEGIN frame would be passed through, but the END frame would get
intercepted to activate a feature. Then, the other conference members would hear
DTMF for forever, which they didn't seem to like very much.
(closes issue #10400, reported by stevefeinstein, fixed by me)
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the ast_check_hangup() funciton. This function takes scheduled hangups into
account.
(closes issue #10230, patch by Juggie)
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few other formatting cleanups, too.
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(closes issue #10277, patches by mvanbaak)
Basically, this changes ...
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3, "Something\n");
to ...
ast_verb(3, "Something\n");
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r77191 | murf | 2007-07-25 16:39:27 -0600 (Wed, 25 Jul 2007) | 1 line
This fix solves problem with intense squelch noise when someone joins conf in bug 9430; We repro'd the problem with meetme opts of 'CciMo'; Josh Colp supplied this patch, and I'm applying it. It looks like playing the recorded username will louse up the next thing played into the channel. Josh rearranged the code so as to start things over before playing data directly into the conference.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r75529 | tilghman | 2007-07-18 07:29:41 -0500 (Wed, 18 Jul 2007) | 2 lines
Using a freed frame causes crashes (closes issue #9317)
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ast_module_user_remove. This is now taken care of in the pbx_exec function outside of the application.
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ast_module_user_hangup_all in it's unload function. The loader will automatically perform this action for it.
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This does not break existing configs - the arguments to p are optional.
Issue 8827, initial patch by junky, mostly rewritten by fw to re-use option p, further modified by me.
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(issue #9984, patches from eliel and dimas)
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* Add support for the reload of sla.conf
(closes issue #9481, patch by me)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r70726 | russell | 2007-06-21 10:21:16 -0500 (Thu, 21 Jun 2007) | 2 lines
Remove a couple of duplicate unlocks
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r69895 | file | 2007-06-19 09:55:25 -0400 (Tue, 19 Jun 2007) | 10 lines
Merged revisions 69894 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r69894 | file | 2007-06-19 09:54:03 -0400 (Tue, 19 Jun 2007) | 2 lines
Perform an extra hangup check just in case. (issue #9589 reported by bcnit)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r69518 | russell | 2007-06-15 10:27:34 -0500 (Fri, 15 Jun 2007) | 5 lines
The SLATRUNK_STATUS variable indicated "SUCCESS" for both an answer of the
incoming call on the trunk, or if the trunk reached its ring timeout.
This patch changes the variable to say "RINGTIMEOUT" in that case.
(issue #9973, reported by n00dle, patch by me)
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(issue #9983, eliel)
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(issue #9957, patches from mvanbaak, caio1982, critch, and dimas)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r69144 | file | 2007-06-13 15:08:24 -0400 (Wed, 13 Jun 2007) | 2 lines
Really ignore NULL frames and check whether the channel hungup or not. (issue #9912 reported by junky)
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guidelines changes
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r67558 | russell | 2007-06-05 18:01:44 -0500 (Tue, 05 Jun 2007) | 5 lines
Fix some crashes related to the use of the "meetme" CLI command. The code for
this command was not locking the conference list at all.
(issue #9351, reported by and patch submitted by Junk-Y, committed patch
is different and by me)
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(Committed from Media Plaza in Utrecht, Netherlands - Open Source VoIP conference)
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is no reason to keep a thread attribute structure on the conference structure.
(Pointed out by Tony Mountifield on the asterisk-dev list)
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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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is actually necessary.
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except it lets you operate on a channel by name instead of conference member
number. It is very useful in combination with the 'X' option to ChanSpy.
(issue #9671, patch by mnicholson, with some small modifications by me)
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