https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r93182 | oej | 2007-12-17 08:15:13 +0100 (MÃ¥n, 17 Dec 2007) | 8 lines
Issue 11574: Add dependencies on res_monitor and res_features.
I wonder if Asterisk can run at all without res_features. My guess is that
there's propably a lot of more modules and the core that depends on it.
Reported by: caio1982
(closes issue #11574)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r93291 | mmichelson | 2007-12-17 13:53:48 -0600 (Mon, 17 Dec 2007) | 6 lines
We need to create the directory for a voicemail user even if they are using IMAP storage
since greetings are stored in the filesystem.
(closes issue #11388, reported by spditner, patch by me inspired by a patch by spditner)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93293 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r93180 | kpfleming | 2007-12-16 22:44:51 -0800 (Sun, 16 Dec 2007) | 23 lines
In http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html,
rizzo brought up some issues related to the way that the metadata required
for menuselect and the rest of the build system is extracted from the source
files. Since I had a few hours to kill on an airplane today, I decided to
improve this situation... so now the system caches the extracted metadata
and uses it to build the menuselect 'tree' as much as it can. The result
of this is that when a single source file is changed, only the metadata for
that file needs to be extracted again, and the rest is used from the cache
files. I also reduced the number of forked processes required to do the
metadata extraction; it was actually possible to do most of what we needed
in the Makefiles themselves without using any shell scripts at all! On my
laptop, these changes resulted in an 80% decrease in the time required
for the 'menuselect.makeopts' automatic check to occur after editing a single
source file.
While doing this work I also cleaned up a few minor things in the Makefiles,
adding a check for 'awk' to the configure script and changed all remaining
places we use 'grep' or 'awk' to use the ones found by the configure script,
and changed the 'prep_tarball' script to build the menuselect metadata so
that tarballs of Asterisk will include it and won't require the user to
wait while it is extracted after unpacking.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93184 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r92323 | mmichelson | 2007-12-11 11:42:25 -0600 (Tue, 11 Dec 2007) | 10 lines
Fixing autofill to be more accurate. Specifically, if calls ahead of the current
caller were ringing members (but not yet bridged) there could be available members
and waiting callers who would not get matched up. The member availability checker
was correctly determining the number of available members in this scenario, but
the queue itself did not parallelly reflect this status on the pending calls. This
commit corrects the issue.
(closes issue #11459, reported by equissoftware, patched by me)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r92202 | mmichelson | 2007-12-10 10:29:44 -0600 (Mon, 10 Dec 2007) | 7 lines
If there are no members in a queue, then the loop where the datastore for detecting
duplicate dialed numbers will be skipped, meaning the datastore isn't created. This means
that when we try to free it, there's a crash. This stops that crash from occurring.
(closes issue #11499, reported by slavon, patched by eliel)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92203 65c4cc65-6c06-0410-ace0-fbb531ad65f3
generate loadable and embedded module lists.
Individual Makefiles now are a lot simpler, possibly as simple as this:
-include $(ASTTOPDIR)/menuselect.makeopts $(ASTTOPDIR)/menuselect.makedeps
MODULE_PREFIX=cdr_
all: _all
include $(ASTTOPDIR)/Makefile.moddir_rules
and also more flexible because in a single directory we can combine
various types of modules (app_, cdr_, func_, ... ) by simply
listing them in the MODULE_PREFIX variable.
The individual Makefiles can also create list of modules to be
excluded by listing them in the variablel MODULE_EXCLUDE (see an
example in channels/Makefile).
With this change it becomes trivial to integrate a directory with
locally created/modified sources into the main build.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r91783 | russell | 2007-12-07 10:38:48 -0600 (Fri, 07 Dec 2007) | 6 lines
* Add channel locking around datastore operations that expect the channel
to be locked.
* Document why we don't record Local channels in the dialed interfaces list.
* Remove the dialed variable as it isn't needed.
* Restructure some code for clarity and coding guidelines stuff
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91784 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r91780 | russell | 2007-12-07 10:25:25 -0600 (Fri, 07 Dec 2007) | 7 lines
* Add channel locking around datastore operations that expect the channel
to be locked.
* Document why we don't record Local channels in the dialed interfaces list.
* Handle memory allocation failure.
* Remove the dialed variable, as it wasn't actually needed.
* Tweak some formatting to conform to coding guidelines.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91781 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r91675 | russell | 2007-12-06 20:19:45 -0600 (Thu, 06 Dec 2007) | 7 lines
Fix in an issue in the call forwarding handling code that was causing crashes
on every call into a queue. I'm not entirely sure about the logic in this part
of the code, so I want to look at it some more tomorrow. However, this makes
it safe and keeps it from crashing.
(closes issue #11486, reported by adamg, patched by me)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Event Dial has new headers, to comply with other events
- Source -> Channel Channel name (caller)
- SrcUniqueID -> UniqueID Uniqueid
(new) -> Dialstring Dialstring in app data
(moremanager)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91407 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r91273 | mmichelson | 2007-12-05 16:35:52 -0600 (Wed, 05 Dec 2007) | 11 lines
The 'G' option for Dial() did not properly handle the case where only a label was
provided. This was due to the fact that the answering channel did not have an extension
set, so ast_parseable_goto would fail. This fix eliminates the call to ast_parseable_goto
on the answering channel since it is a wasteful call. The answering channel and the calling
channel are both directed to the same extension and context, just different priorities, so
we can just copy the values from the calling channel to the answering channel and increment
the answering channel's priority.
(closes issue #11382, reported by jon, patch by me with correction by jon)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91291 65c4cc65-6c06-0410-ace0-fbb531ad65f3
does not modify the contents of the "mailbox" string. In other words, I'm changing
the imap_retrieve_file function to take a const char* as the third argument so that I
don't need to cast const char*'s as char*'s to suppress compiler warnings.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90930 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r90470 | russell | 2007-12-02 12:18:52 -0600 (Sun, 02 Dec 2007) | 6 lines
The other day when I went through making changes as a result of the ao2_link()
change, I added some code to set pointers to NULL after they were unreferenced.
This pointed out that in this place, the object was unreferenced before the
code was done using it. So, move the unref down a little bit.
(crash reported by jmls on IRC)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90471 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90388 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r90348 | russell | 2007-11-30 13:26:04 -0600 (Fri, 30 Nov 2007) | 8 lines
Change the behavior of ao2_link(). Previously, in inherited a reference.
Now, it automatically increases the reference count to reflect the reference
that is now held by the container.
This was done to be more consistent with ao2_unlink(), which automatically
releases the reference held by the container. It also makes it so it is
no longer possible for a pointer to be invalid after ao2_link() returns.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90351 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r90163 | mmichelson | 2007-11-29 13:38:39 -0600 (Thu, 29 Nov 2007) | 6 lines
This patch handles the case where a queue member with a negative penalty is added
via the manager. If a negative value is submitted for a member penalty, we set it to 0.
(closes issue #11411, reported and patched by Laureano)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90164 65c4cc65-6c06-0410-ace0-fbb531ad65f3
the diff to trunk.
This just removes some checks on the return value of alloca(), as behavior
is undefined if it runs out of stack space, and we don't check it anywhere else.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r89837 | mmichelson | 2007-11-27 17:10:05 -0600 (Tue, 27 Nov 2007) | 12 lines
Two changes with regards to the 'eventwhencalled' option of queues.conf
1) Due to some signed vs. unsigned silliness, setting 'eventwhencalled' to
'vars' or 'yes' did exactly the same thing. Thus the sign change of the
ast_true call.
2) The vars2manager function overwrote a \n for every channel variable it parsed, resulting
in bizarre output for the channel variables. This patch remedies this.
(related to issue #11385, however I'm not sure if this will actually be enough to close it)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r89622 | murf | 2007-11-26 23:24:02 -0700 (Mon, 26 Nov 2007) | 1 line
closes issue #11379; OK, this is an attempt to make both sides happy. To the cdr.conf file, I added the option 'unanswered', which defaults to 'no'. In this mode, you will see a cdr for a call, whether it was answered or not. The disposition will be NO ANSWER or ANSWERED, as appropriate. The src is as you'd expect, the destination channel will be one of the channels from the Dial() call, usually the last in the list if more than one chan was specified. With unanswered set to 'yes', you will still see this cdr entry in both cases. But in the case where the dial timed out, you will also see a cdr for each line attempted, marked NO ANSWER, with no destination channel name. The new option defaults to 'no', so you don't see the pesky extra cdr's by default, and you will not see the irritating 'not posted' messages.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r89618 | mmichelson | 2007-11-26 17:10:49 -0600 (Mon, 26 Nov 2007) | 7 lines
After issuing a "say load new", if a caller hangs up during the middle of playback of a number,
app_playback will continue to try to play the remaining files. With this change, no more files will
be played back upon hangup.
(closes issue #11345, reported and patched by IgorG)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89619 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Restructure other changes to UPGRADE.txt and CHANGES
We're still looking for scripts that replace
asterisk -rx "show shannels concise"
by using the manager interface, but still produces the same output.
Anyone?
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r89580 | mmichelson | 2007-11-26 09:48:06 -0600 (Mon, 26 Nov 2007) | 6 lines
Revert vmu->email back to an empty string if it was empty when imap_store_file
was called. This prevents sending a duplicate e-mail.
(closes issue #11204, reported by spditner, patched by me)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89581 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r89540 | tilghman | 2007-11-24 00:19:23 -0600 (Sat, 24 Nov 2007) | 9 lines
Currently, zero-length voicemail messages cause a hangup in VoicemailMain.
This change fixes the problem, with a multi-faceted approach. First, we
do our best to avoid these messages from being created in the first place,
and second, if that fails, we detect when the voicemail message is
zero-length and avoid exiting at that point.
Reported by: dtyoo
Patch by: gkloepfer,tilghman
(Closes issue #11083)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89522 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r89493 | mmichelson | 2007-11-21 13:24:22 -0600 (Wed, 21 Nov 2007) | 5 lines
Changing an inaccurate debug message to be less inaccurate. Under the circumstances, this
message would always report that there were 0 members available, even though that may not be true.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89494 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Added the ability to specify the music on hold class used to play into the
conference when there is only one member and the M option is used.
* Added the ability to specify a music on hold class to play instead of ringing
for the SLATrunk application.
(patched by me, and tested internally)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89470 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After this commit we can actually load modules under windows,
and we can start debugging more interesting problems related
to the load order and functionality of modules.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3
to play the name(s) of the person(s) to whom you are forwarding the message prior to
prompting for prepending. If no name is found, the extension is read back verbatim.
(closes issue #9046, reported and patched by jaroth)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89297 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89293 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This prevents modifying the strings in the stored variables,
and catched a few instances where this was actually done.
Given the differences between trunk and 1.4 (and the fact that this
is effectively an API change) it is better to fix 1.4 independently.
These are
chan_sip.c::sip_register()
chan_skinny.c:: near line 2847
config.c:: near line 1774
logger.c::make_components()
res_adsi.c:: near line 1049
I may have missed some instances for modules that do not build here.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r89205 | mmichelson | 2007-11-12 18:56:46 -0600 (Mon, 12 Nov 2007) | 5 lines
Some sanity checking for MixMonitor. If only 1 argument is given, then the args.options
and args.post_process strings are uninitialized and could contain garbage. This change
handles this situation properly by only using arguments that we have parsed.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89206 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- 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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89106 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r89093 | tilghman | 2007-11-07 17:39:37 -0600 (Wed, 07 Nov 2007) | 7 lines
The member refcount must be incremented, to avoid using it after deallocation.
A huge thanks go to lvl- for patiently providing the necessary valgrind output
that was necessary to finding this problem of memory corruption.
Reported by: lvl-
Patch by: tilghman
Closes issue #11174
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89094 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89073 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87971 65c4cc65-6c06-0410-ace0-fbb531ad65f3
phones to be rung in a specific order.
(closes issue #7279, reported and initially patched by diLLec, patch reworked by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87154 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87040 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r86328 | mmichelson | 2007-10-18 12:38:26 -0500 (Thu, 18 Oct 2007) | 5 lines
If a non-existent file is specified to be played either as a periodic announcement
or as a hold/position announcement, the caller would be kicked out of the queue.
No longer does this happen.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r86202 | mmichelson | 2007-10-17 16:39:05 -0500 (Wed, 17 Oct 2007) | 6 lines
Changing the strategy field of the call_queue struct to be signed instead of unsigned,
since the code attempts to set the strategy to -1 if you specify a bogus strategy.
While this isn't a huge issue in 1.4, it could be a problem for someone who, say, tries
to use the roundrobin strategy in trunk (despite all the deprecation warnings in 1.4).
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86208 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r85958 | mmichelson | 2007-10-16 16:14:34 -0500 (Tue, 16 Oct 2007) | 5 lines
Trying to remove a non-dynamic queue member via dynamic means can lead to some
interesting (read nasty) situations. This patch clears up the issue by making
only dynamic queue members removable via dynamic methods.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85959 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Many uses of the astlisting environment around verbatim text to ensure that
it gets properly formatted and doesn't run off the page.
- Update some things that have been deprecated.
- Add escaping as needed
- and more ...
(closes issue #10978)
Reported by: IgorG
Patches:
texdoc-85542-1.patch uploaded by IgorG (license 20)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85547 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r85242 | file | 2007-10-10 11:14:56 -0300 (Wed, 10 Oct 2007) | 6 lines
Close voicemail message description file if duration did not meet the minimum, or else we will eventually run out of file descriptors.
(closes issue #10918)
Reported by: brak2718
Patches:
vm1.4.12.1.patch uploaded by brak2718 (license 279)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85254 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1) Fix some bad logic in the counting of statistics for QueueSummary manager event. Variables were not being
reset for each additional queue, so cumulative totals were reported on each successive queue.
2) Add a longest hold time stat to QueueSummary manager event.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@84726 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@84173 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r83179 | russell | 2007-09-19 14:50:48 -0500 (Wed, 19 Sep 2007) | 5 lines
The System() and TrySystem() applications can take a substantial amount of
time to execute while not servicing the channel. So, put the channel in
autoservice while the command is being executed.
(closes issue #10726, reported by mnicholson)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@83180 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r83074 | file | 2007-09-19 10:47:59 -0300 (Wed, 19 Sep 2007) | 6 lines
Protect the CDR record from modification by pbx_exec so that the application data contains the Queue data.
(closes issue #10761)
Reported by: snar
Patches:
app-queue-mixmonitor.patch uploaded by snar (license 245)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@83075 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r82992 | russell | 2007-09-18 19:19:49 -0500 (Tue, 18 Sep 2007) | 4 lines
Change the description of app_flash to note how it can be a useful tool instead
of just saying that it is generally a worthless feature.
(Thanks to Jim Van Meggelen for pointing it out and providing the proposed text)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82993 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: ruffle
Patches:
app_voicemail.c.diff uploaded by ruffle (license 201)
10739-moveheard.diff uploaded by qwell (license 4)
Tested by: callguy, ruffle
Add an option to disable the automatic moving of "heard" messages to the Old folder.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82871 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r82865 | mmichelson | 2007-09-18 15:09:02 -0500 (Tue, 18 Sep 2007) | 4 lines
Moving the logic for handling an empty membername to the create_member function so that there is a common place
where this occurs instead of being spread out to several different places.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82866 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: snar
Patches:
app-queue-cdr-trunk.patch uploaded by snar (license 245)
queues.conf.patch uploaded by snar (license 245)
Add an updatecdr option to queues.conf, so that if a "member name" is specified,
the cdr record will be updated with that, rather than the channel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82800 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r82644 | russell | 2007-09-17 15:00:32 -0500 (Mon, 17 Sep 2007) | 6 lines
Initialize some memory to fix crashes when leaving voicemail. This problem
was fixed by running Asterisk under valgrind.
(closes issue #10746, reported by arcivanov, patched by me)
*** IMPORTANT NOTE: We need to check to see if this same bug exists elsewhere.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82660 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r82590 | kpfleming | 2007-09-17 11:33:30 -0500 (Mon, 17 Sep 2007) | 2 lines
fix a couple of places where a logical member name (if specified) was not used, but instead the direct interface was listed
........
r82592 | kpfleming | 2007-09-17 11:40:12 -0500 (Mon, 17 Sep 2007) | 2 lines
revert a change that wasn't supposed to be committed... doh!
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82593 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r82326 | mmichelson | 2007-09-13 11:25:59 -0500 (Thu, 13 Sep 2007) | 7 lines
Added logic to handle the unlikely case that someone has two queues with the same name.
Asterisk will log a warning message letting the user know that one was already defined with that
name and is it skipping all further instances. This also will work for realtime queues but in order
for that to happen, the user would have to trigger a perfectly timed reload as a realtime queue is being
looked up, which is highly unlikely (but taken care of nonetheless).
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r82274 | mmichelson | 2007-09-12 09:24:53 -0500 (Wed, 12 Sep 2007) | 6 lines
We should only initialize a realtime queue when it is allocated, not every time we access it. This prevents the members ao2_container
from being reallocated every time the queue is accessed.
I also removed a debug message I had accidentally left in on a previous commit.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r82263 | russell | 2007-09-11 15:49:34 -0500 (Tue, 11 Sep 2007) | 5 lines
Fix another missing unref of member objects. This one was pointed out by Marta.
When building the outgoing list in try_calling(), a member reference is stored
in each outgoing entry. However, when this list got destroyed, the reference
was not released.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82264 65c4cc65-6c06-0410-ace0-fbb531ad65f3