Kevin P. Fleming [Tue, 4 Dec 2007 17:40:29 +0000 (17:40 +0000)]
fix build of this module when libpri and/or libss7 are or are not present
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90880
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Jason Parker [Tue, 4 Dec 2007 17:38:51 +0000 (17:38 +0000)]
Merged revisions 90876 via svnmerge from
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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Russell Bryant [Tue, 4 Dec 2007 17:36:59 +0000 (17:36 +0000)]
Fix a silly little typo :)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90878
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Jason Parker [Tue, 4 Dec 2007 17:35:40 +0000 (17:35 +0000)]
Fix build in trunk. This was fixed in 1.4, but blocked in trunk since this hadn't been merged yet.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90877
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Mark Michelson [Tue, 4 Dec 2007 17:08:36 +0000 (17:08 +0000)]
Merged revisions 90735 via svnmerge from
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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Olle Johansson [Tue, 4 Dec 2007 15:16:03 +0000 (15:16 +0000)]
(closes issue #11431)
Reported by: Laureano
Patches:
app_queue.c.patch uploaded by Laureano (license 265)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90854
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Olle Johansson [Tue, 4 Dec 2007 15:07:53 +0000 (15:07 +0000)]
(closes issue #11422)
Reported by: eliel
Patches:
core.show.hint.patch uploaded by eliel (license 64)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90853
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Olle Johansson [Tue, 4 Dec 2007 15:02:48 +0000 (15:02 +0000)]
(closes issue #11462)
Reported by: eliel
Patches:
CHANGES.patch uploaded by eliel (license 64)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90852
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Tilghman Lesher [Tue, 4 Dec 2007 15:01:57 +0000 (15:01 +0000)]
Pass the Asterisk version to AGI scripts as part of the initial dump of info
Reported by: acunningham
Patch by: acunningham
(Closes issue #11398)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90851
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Luigi Rizzo [Tue, 4 Dec 2007 11:50:05 +0000 (11:50 +0000)]
fix build on cygwin
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90834
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Joshua Colp [Tue, 4 Dec 2007 05:31:00 +0000 (05:31 +0000)]
Blocked revisions 90798 via svnmerge
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r90798 | file | 2007-12-04 01:29:33 -0400 (Tue, 04 Dec 2007) | 2 lines
Fix build issue on the build cluster.
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Tilghman Lesher [Mon, 3 Dec 2007 23:52:23 +0000 (23:52 +0000)]
Merged revisions 90753 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r90753 | tilghman | 2007-12-03 17:50:51 -0600 (Mon, 03 Dec 2007) | 5 lines
Solaris requires the inclusion of sys/loadavg.h for getloadavg().
Reported by: snuffy
Patch by: snuffy,tilghman
(Closes issue #11430)
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Steve Murphy [Mon, 3 Dec 2007 23:49:40 +0000 (23:49 +0000)]
A small fix from snuffy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90746
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Jason Parker [Mon, 3 Dec 2007 23:48:04 +0000 (23:48 +0000)]
Add manager events for when a monitor is started or stopped.
Closes issue #10191, patch by dgradecak.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90738
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Tilghman Lesher [Mon, 3 Dec 2007 23:29:57 +0000 (23:29 +0000)]
Merged revisions 90736 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r90736 | tilghman | 2007-12-03 17:23:55 -0600 (Mon, 03 Dec 2007) | 5 lines
If both dbhost and dbsock were not set, a NULL deref could result
Reported by: xrg
Patch by: tilghman
(Closes issue #11387)
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Jason Parker [Mon, 3 Dec 2007 22:07:57 +0000 (22:07 +0000)]
Merged revisions 90696 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
(Closes issue #11383)
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r90696 | qwell | 2007-12-03 16:06:36 -0600 (Mon, 03 Dec 2007) | 4 lines
Make sure we always close the conference fd if we have an open one.
Issue 11383, reported by markmhy, patch by eliel.
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Mark Michelson [Mon, 3 Dec 2007 21:24:56 +0000 (21:24 +0000)]
Replacing some calls to free() with ast_free().
(closes issue #11448, reported and patched by jaroth)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90670
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Joshua Colp [Mon, 3 Dec 2007 21:03:05 +0000 (21:03 +0000)]
Add AGI commands for speech recognition. These mirror the dialplan applications mostly but present the information in a nicer fashion. The SPEECH RECOGNIZE command for example will return the results instead of having to query the dialplan functions.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90656
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Mark Michelson [Mon, 3 Dec 2007 21:00:44 +0000 (21:00 +0000)]
Merged revisions 90639 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r90639 | mmichelson | 2007-12-03 14:59:51 -0600 (Mon, 03 Dec 2007) | 5 lines
Changing some bad logic when calculating the interdigit timeout.
(closes issue #11402, reported and patched by eferro)
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Jason Parker [Mon, 3 Dec 2007 20:58:26 +0000 (20:58 +0000)]
Merged revisions 90607 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
(Closes issue #11436)
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r90607 | qwell | 2007-12-03 14:51:17 -0600 (Mon, 03 Dec 2007) | 4 lines
Fix crash in ParkAndAnnounce application.
Issue #11436, reported by lytledd, patch by eliel.
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Tilghman Lesher [Mon, 3 Dec 2007 20:30:02 +0000 (20:30 +0000)]
Avoid an additional function call. Reported by: eliel Patch by: eliel (Closes issue #11438)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90591
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Joshua Colp [Mon, 3 Dec 2007 20:07:34 +0000 (20:07 +0000)]
Merged revisions 90588 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r90588 | file | 2007-12-03 16:05:42 -0400 (Mon, 03 Dec 2007) | 2 lines
Do not create a smoother for G723.1 frames, they need to be left alone to their native 20/24 byte size.
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Joshua Colp [Mon, 3 Dec 2007 18:44:16 +0000 (18:44 +0000)]
Merged revisions 90548 via svnmerge from
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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Jason Parker [Mon, 3 Dec 2007 18:21:49 +0000 (18:21 +0000)]
Blocked revisions 90546 via svnmerge
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r90546 | qwell | 2007-12-03 12:20:49 -0600 (Mon, 03 Dec 2007) | 4 lines
Only log debug messages if debug is enabled.
Closes issue #11416, patch by casper.
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Mark Michelson [Mon, 3 Dec 2007 16:46:01 +0000 (16:46 +0000)]
Updating sample queues.conf file to show how multiple periodic announcements
may be specified since this was not documented previously
(closes issue #11432, reported and patched by Laureano)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90528
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Joshua Colp [Mon, 3 Dec 2007 14:14:43 +0000 (14:14 +0000)]
Remove the file descriptors from the main poll channel when the channel is hung up during the dialing attempt, and make sure a channel exists before trying to remove it at the end.
(closes issue #11441)
Reported by: blitzrage
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90508
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Russell Bryant [Sun, 2 Dec 2007 18:20:13 +0000 (18:20 +0000)]
Merged revisions 90470 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90471
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Tilghman Lesher [Sun, 2 Dec 2007 09:42:48 +0000 (09:42 +0000)]
Merged revisions 90432 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r90432 | tilghman | 2007-12-02 03:34:23 -0600 (Sun, 02 Dec 2007) | 7 lines
Clarify the return value on autoservice. Specifically, if you started
autoservice and autoservice was already on, it would erroneously return an
error.
Reported by: adiemus
Patch by: dimas
(Closes issue #11433)
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Jason Parker [Sat, 1 Dec 2007 01:37:50 +0000 (01:37 +0000)]
Only reload if the config file has changed.
Closes issue #11281, patch by eliel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90410
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Mark Michelson [Fri, 30 Nov 2007 21:19:57 +0000 (21:19 +0000)]
Adding support for the "automixmonitor" dial and queue options.
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
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Russell Bryant [Fri, 30 Nov 2007 19:34:47 +0000 (19:34 +0000)]
Merged revisions 90348 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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Russell Bryant [Fri, 30 Nov 2007 18:47:14 +0000 (18:47 +0000)]
Merged revisions 90310 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r90310 | russell | 2007-11-30 12:46:46 -0600 (Fri, 30 Nov 2007) | 2 lines
Add some notes on the behavior of ao2_unlink() after a discussion with Tilghman
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Joshua Colp [Fri, 30 Nov 2007 14:45:36 +0000 (14:45 +0000)]
Merged revisions 90269 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r90269 | file | 2007-11-30 10:43:15 -0400 (Fri, 30 Nov 2007) | 6 lines
Fix locking issues under one legged replaces scenarios.
(closes issue #11420)
Reported by: irroot
Patches:
chan_sip_oneleg.patch uploaded by irroot (license 52)
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Mark Michelson [Fri, 30 Nov 2007 00:16:45 +0000 (00:16 +0000)]
Merged revisions 90231 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r90231 | mmichelson | 2007-11-29 18:16:04 -0600 (Thu, 29 Nov 2007) | 5 lines
Clear the DTMF buffer if the call times out.
(closes issue #11418, reported and patched by eferro)
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Mark Michelson [Thu, 29 Nov 2007 19:39:31 +0000 (19:39 +0000)]
Merged revisions 90163 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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Tilghman Lesher [Thu, 29 Nov 2007 19:35:49 +0000 (19:35 +0000)]
Merged revisions 90160 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r90160 | tilghman | 2007-11-29 13:24:11 -0600 (Thu, 29 Nov 2007) | 2 lines
Properly escape input buffers (Fixes AST-2007-025)
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Tilghman Lesher [Thu, 29 Nov 2007 17:50:44 +0000 (17:50 +0000)]
Merged revisions 90155 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r90155 | tilghman | 2007-11-29 11:29:59 -0600 (Thu, 29 Nov 2007) | 5 lines
Use of "private" as a field name in a header file messes with C++ projects
Reported by: chewbacca
Patch by: casper
(Closes issue #11401)
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Tilghman Lesher [Thu, 29 Nov 2007 17:42:21 +0000 (17:42 +0000)]
Fix build of trunk
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90157
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Tilghman Lesher [Thu, 29 Nov 2007 17:31:35 +0000 (17:31 +0000)]
Merged revisions 90154 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r90154 | tilghman | 2007-11-29 11:18:09 -0600 (Thu, 29 Nov 2007) | 2 lines
Upgrade the core sounds release version
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Kevin P. Fleming [Thu, 29 Nov 2007 13:38:11 +0000 (13:38 +0000)]
let's try this again... *all* compilation and linking in Asterisk should be done using the standard compilation rules, not manually created ones. changing hashtest.c to use these rules caused the compiler to notice a large number of coding guidelines violations, so those are fixed too.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90150
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Kevin P. Fleming [Thu, 29 Nov 2007 13:30:50 +0000 (13:30 +0000)]
restore behavior from the 1.4 branch... manager users created via users.conf should default to *all* permissions, not none
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90149
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Russell Bryant [Thu, 29 Nov 2007 00:37:22 +0000 (00:37 +0000)]
Blocked revisions 90147 via svnmerge
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r90147 | russell | 2007-11-28 18:36:59 -0600 (Wed, 28 Nov 2007) | 1 line
fix some formatting i accidentally changed
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Russell Bryant [Thu, 29 Nov 2007 00:28:10 +0000 (00:28 +0000)]
Merged revisions 90145 via svnmerge from
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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Russell Bryant [Thu, 29 Nov 2007 00:07:32 +0000 (00:07 +0000)]
Merged revisions 90142 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r90142 | russell | 2007-11-28 18:06:08 -0600 (Wed, 28 Nov 2007) | 4 lines
Merge a change from team/russell/chan_refcount ...
This makes ast_stopstream() thread-safe.
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Russell Bryant [Wed, 28 Nov 2007 23:57:30 +0000 (23:57 +0000)]
Merge another small doxygen change from team/russell/chan_refcount to indicate
that a channel doesn't need to be locked before calling a certain function.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90141
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Russell Bryant [Wed, 28 Nov 2007 23:47:26 +0000 (23:47 +0000)]
Merge some channel.h doxygen updates from team/russell/chan_refcount
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90139
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Joshua Colp [Wed, 28 Nov 2007 23:03:09 +0000 (23:03 +0000)]
Merged revisions 90101 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r90101 | file | 2007-11-28 18:59:28 -0400 (Wed, 28 Nov 2007) | 6 lines
Fix a few memory leaks.
(closes issue #11405)
Reported by: eliel
Patches:
load_realtime.patch uploaded by eliel (license 64)
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Kevin P. Fleming [Wed, 28 Nov 2007 22:44:38 +0000 (22:44 +0000)]
Merged revisions 90098 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r90098 | kpfleming | 2007-11-28 16:30:46 -0600 (Wed, 28 Nov 2007) | 2 lines
it is impossible to set permissions for manager accounts created by users.conf (reported internally, patched by me)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90100
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Joshua Colp [Wed, 28 Nov 2007 22:32:57 +0000 (22:32 +0000)]
file says... compile before you commit!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90099
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Mark Michelson [Wed, 28 Nov 2007 22:17:38 +0000 (22:17 +0000)]
Removing a pointless check of option_debug
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90061
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Mark Michelson [Wed, 28 Nov 2007 22:10:36 +0000 (22:10 +0000)]
Merged revisions 90059 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r90059 | mmichelson | 2007-11-28 16:08:50 -0600 (Wed, 28 Nov 2007) | 13 lines
Removing some seemingly pointless code. This sets a channel variable for every priority
executed in the dialplan if you have debug set to anything non-zero. This seems pointless
due to the fact that these channel variables are not referenced anywhere else in the code and
their names are esoteric enough that they would not be practical to reference in the dialplan. Plus
the fact that this behavior isn't documented anywhere means that the change is not likely to cause
any disruption. If anything, this may actually cause a slight performance increase if running with
debug on.
The motivating influence for this code change is the eventwhencalled option for queues. If set to
vars, all channel variables will be output to the manager. These unnecessary channel variables make
the output a lot more difficult to deal with.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90060
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Steve Murphy [Wed, 28 Nov 2007 20:33:21 +0000 (20:33 +0000)]
Made expr2 parser 8-bit transparent
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90039
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Jason Parker [Wed, 28 Nov 2007 20:27:40 +0000 (20:27 +0000)]
Remove "old"-style CLI handler, since nothing uses it anymore.
Closes issue #11403, patch by eliel. This also completes the janitor project.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90038
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Mark Michelson [Wed, 28 Nov 2007 17:37:18 +0000 (17:37 +0000)]
Blocked revisions 89999 via svnmerge
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r89999 | mmichelson | 2007-11-28 11:30:47 -0600 (Wed, 28 Nov 2007) | 6 lines
Recording greetings when using IMAP storage was causing zero-length files to be stored.
Since greetings are not retrieved from IMAP anyway, it is pointless to attempt storing them there.
(closes issue #11359, reported by spditner, patched by me)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90000
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Joshua Colp [Wed, 28 Nov 2007 15:48:00 +0000 (15:48 +0000)]
Hide CLI commands starting with _ from tab completion as was done previously.
(closes issue #11395)
Reported by: eliel
Patches:
cli.c.patch uploaded by eliel (license 64)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89982
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Joshua Colp [Wed, 28 Nov 2007 15:38:03 +0000 (15:38 +0000)]
Fix a few log messages.
(closes issue #11396)
Reported by: IgorG
Patches:
spell.v1.diff uploaded by IgorG (license 20)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89981
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Russell Bryant [Wed, 28 Nov 2007 00:49:55 +0000 (00:49 +0000)]
Merge some little changes from team/russell/chan_refcount to help reduce
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
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Mark Michelson [Wed, 28 Nov 2007 00:47:22 +0000 (00:47 +0000)]
Adding support for realtime music on hold. The following are the main points:
1. When moh is started, we search first in memory to find the class. If we do not
find it in memory, we search realtime instead.
2. When moh is restarted (as in, it had been started on this particular channel, stopped,
and now we're starting it again), if using the "files" mode, then realtime will always
be rechecked. If you are using other modes, however, we will simply reattach to the external
running process which was playing moh earlier in the call. This is a necessary compromise so that
we don't end up with too many background processes.
3. musiconhold.conf has a general section now. It has one option: cachertclasses. If set to yes,
then moh classes found in realtime will be added to the in-memory list. This has the advantage
of not requiring database lookups each time moh is started, but it has the disadvantage of not
truly being realtime.
I have tested this for functionality, and it passes. I also tested this under valgrind and there
are no memory problems reported under typical use.
Special thanks to Sergee for implementing this feature and enduring my complaints on the bugtracker!
(closes issue #11196, reported and patched by sergee)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89946
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Russell Bryant [Wed, 28 Nov 2007 00:24:19 +0000 (00:24 +0000)]
Merged revisions 89893 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89893 | russell | 2007-11-27 18:20:13 -0600 (Tue, 27 Nov 2007) | 4 lines
- update documentation for some of the goto functions to note that they
handle locking the channel as needed
- update ast_explicit_goto() to lock the channel as needed
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Russell Bryant [Tue, 27 Nov 2007 23:56:45 +0000 (23:56 +0000)]
Document that the channel is not locked when the send_digit_begin and end
callbacks get called.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89891
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Russell Bryant [Tue, 27 Nov 2007 23:50:58 +0000 (23:50 +0000)]
Merged revisions 89886 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89886 | russell | 2007-11-27 17:47:28 -0600 (Tue, 27 Nov 2007) | 2 lines
Don't do frame processing if ast_read() returned NULL.
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Russell Bryant [Tue, 27 Nov 2007 23:50:25 +0000 (23:50 +0000)]
Merge changes from team/russell/iax2_frame_queue
This patch is an optimization for chan_iax2. This module is now heavily
multi-threaded. However, there is still a good number of globally shared
resources that prevent things from happen asynchronously. One of those things
was the global IAX frame queue. This queue was used to hold frames that have
been deferred for transmitting by another thread, and frames that may need to
get retransmitted.
I changed the frame queue to be per-call, since almost all of the frame queue
handling only cares about frames specific to a call number.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89887
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Russell Bryant [Tue, 27 Nov 2007 23:21:38 +0000 (23:21 +0000)]
Merged revisions 89844 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89844 | russell | 2007-11-27 17:21:13 -0600 (Tue, 27 Nov 2007) | 3 lines
Instead of depending on the return value of ast_true(), explicitly set the
eventwhencalled variable to 1.
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Russell Bryant [Tue, 27 Nov 2007 23:17:36 +0000 (23:17 +0000)]
Merged revisions 89839 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89839 | russell | 2007-11-27 17:16:00 -0600 (Tue, 27 Nov 2007) | 2 lines
Don't start/stop autoservice in pbx_extension_helper() unless a channel exists
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Mark Michelson [Tue, 27 Nov 2007 23:11:12 +0000 (23:11 +0000)]
Merged revisions 89837 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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Russell Bryant [Tue, 27 Nov 2007 22:42:57 +0000 (22:42 +0000)]
Bring in a small change from team/russell/chan_refcount
This replaces tab completion code with the use of a public function that
does the same thing
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89835
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Steve Murphy [Tue, 27 Nov 2007 22:14:55 +0000 (22:14 +0000)]
closes issue #11294; missed the conditional unlock of the contexts when the hash table is used instead; also, used the ast_free_ptr as advised.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89792
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Russell Bryant [Tue, 27 Nov 2007 22:05:36 +0000 (22:05 +0000)]
Merged revisions 89790 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89790 | russell | 2007-11-27 15:45:51 -0600 (Tue, 27 Nov 2007) | 41 lines
Merge changes from team/russell/autoservice_1.4
This set of changes fixes an issue that was reported to me on IRC yesterday.
The user, d1mas, was using chan_zap for incoming calls and was having DTMF
recognition issues in some situations. Specifically, he noticed that the
problem occurred when using DISA or WaitExten. He also noticed that when
using Read, the problem did not occur. His system also used DUNDi for
dialplan lookups.
So, he theorized that if the DUNDi lookups blocked for some period of time,
that audio from the zap channel could get lost. If the audio got lost, then
it wouldn't be run through the DTMF detector, and digits could get lost.
He was correct, and the following set of changes fixes the problem. However,
the changes go a little bit further than what was necessary to fix this exact
problem.
1) I updated pbx_extension_helper() to autoservice the associated channel to
handle cases where extension lookups may take a long time. This would
normally be a dialplan switch that does some lookup over the network, such
as the DUNDi or IAX2 switches.
This ensures that even while a DUNDi lookup is blocking, the channel will be
continuously serviced.
2) I made a change to the autoservice code. This is actually something that
has bothered me for a long time. When a channel is in autoservice, _all_
frames get thrown away. However, some frames really shouldn't be thrown
away. The most notable examples are signalling (CONTROL) frames, and DTMF.
So, this patch queues up important frames while a channel is in autoservice.
When autoservice is stopped on the channel, the queued up frames get stuck
back on the channel so that they can get processed instead of thrown away.
3) I made another change to the autoservice code to handle the case where
autoservice is started on channels recursively.
Previously, you could call ast_autoservice_start() multiple times on a
channel, and it would stop the first time ast_autoservice_stop() gets
called. Now, it will ensure that autoservice doesn't actually stop until
the final call to ast_autoservice_stop().
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Olle Johansson [Tue, 27 Nov 2007 21:10:50 +0000 (21:10 +0000)]
A few more "moremanager" fixes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89772
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Olle Johansson [Tue, 27 Nov 2007 21:04:29 +0000 (21:04 +0000)]
More "moremanager" fixes. Manager commands to check module status.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89771
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Olle Johansson [Tue, 27 Nov 2007 20:50:48 +0000 (20:50 +0000)]
More "moremanager" changes - doxygen docs and changing manager version (finally)
before making more dramatic changes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89770
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Olle Johansson [Tue, 27 Nov 2007 20:36:59 +0000 (20:36 +0000)]
More additions from the "moremanager" branch, this time for IAX2.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89769
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Mark Michelson [Tue, 27 Nov 2007 20:24:08 +0000 (20:24 +0000)]
Blocked revisions 89727 via svnmerge
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r89727 | mmichelson | 2007-11-27 14:22:59 -0600 (Tue, 27 Nov 2007) | 6 lines
Changing some calls from free() to ast_free() since they were allocated with
ast_calloc().
(closes issue #11390, reported and patched by Laureano)
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Kevin P. Fleming [Tue, 27 Nov 2007 20:21:57 +0000 (20:21 +0000)]
Merged revisions 89709 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89709 | kpfleming | 2007-11-27 14:16:56 -0600 (Tue, 27 Nov 2007) | 2 lines
on second thought... revert all the other changes i've made in app options parsing leaving only one: if an empty argument is supplied for an option, set that argument pointer to point to an empty string rather than NULL, so that the application can do normal checks on it without worrying about it being NULL
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Russell Bryant [Tue, 27 Nov 2007 20:17:36 +0000 (20:17 +0000)]
remove a duplicate manager event
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89710
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Olle Johansson [Tue, 27 Nov 2007 19:50:12 +0000 (19:50 +0000)]
Manager events from the "moremanager" branch
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89706
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Kevin P. Fleming [Tue, 27 Nov 2007 19:47:19 +0000 (19:47 +0000)]
Merged revisions 89701 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89701 | kpfleming | 2007-11-27 13:36:55 -0600 (Tue, 27 Nov 2007) | 2 lines
generate a warning when an application option that requires an argument is ignored due to lack of an argument
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Olle Johansson [Tue, 27 Nov 2007 19:45:39 +0000 (19:45 +0000)]
Starting to merge changes from the "moremanager" branch. Documentation will
follow.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89702
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Olle Johansson [Tue, 27 Nov 2007 19:24:17 +0000 (19:24 +0000)]
The following patch with updates for trunk. Works much better in trunk.
Also by accident fixed a bad typo by a previous committer, which actually made video calls
not work fully...
Merged revisions 89630 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89630 | oej | 2007-11-27 16:23:17 +0100 (Tis, 27 Nov 2007) | 12 lines
If we get a codec offer using a well-known payload type, but using it for another
codec that we don't know, Asterisk did not remove that codec from the list.
With this patch, we remove the codec from audio and video rtp objects and
deny it ever existed. Thanks to lasse for testing.
(closes issue #11376)
Reported by: lasse
Patches:
bug11376.txt uploaded by oej (license 306)
Tested by: lasse
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Jason Parker [Tue, 27 Nov 2007 19:12:33 +0000 (19:12 +0000)]
Add an S_COR macro, which is similar to the existing S_OR macro,
except with an additional boolean arg.
A hack such as:
foo ? S_OR(bar, "baz") : "baz"
becomes:
S_COR(foo, bar, "baz")
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89683
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Steve Murphy [Tue, 27 Nov 2007 18:50:44 +0000 (18:50 +0000)]
made AEL 8-bit transparent; mainly the lexer was tossing chars with the hi-order bit set. Not nice. Also, allow @ in extension names, and a backslash, also.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89682
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Joshua Colp [Tue, 27 Nov 2007 17:01:19 +0000 (17:01 +0000)]
Ensure the value returned from ast_random is between 0 and RAND_MAX on 64-bit platforms.
(closes issue #11348)
Reported by: sperreault
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89637
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Russell Bryant [Tue, 27 Nov 2007 16:13:14 +0000 (16:13 +0000)]
Merged revisions 89634 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89634 | russell | 2007-11-27 10:12:33 -0600 (Tue, 27 Nov 2007) | 3 lines
Add a note to the sample voicemail config noting that when using IMAP storage,
only the first format specified will be attached to the message.
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Tilghman Lesher [Tue, 27 Nov 2007 15:41:46 +0000 (15:41 +0000)]
Merged revisions 89631 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89631 | tilghman | 2007-11-27 09:38:03 -0600 (Tue, 27 Nov 2007) | 3 lines
Default result of STAT should be "0" not "".
Reported via the -users mailing list, fixed by me.
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Olle Johansson [Tue, 27 Nov 2007 07:36:54 +0000 (07:36 +0000)]
Merged revisions 89624 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89624 | oej | 2007-11-27 08:34:19 +0100 (Tis, 27 Nov 2007) | 6 lines
Clarify limitonpeers=yes
(closes issue #11304)
Reported by: pj
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Steve Murphy [Tue, 27 Nov 2007 06:47:08 +0000 (06:47 +0000)]
Merged revisions 89622 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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Mark Michelson [Mon, 26 Nov 2007 23:15:53 +0000 (23:15 +0000)]
Change all instances of "CALLERID(number)" to "CALLERID(num)" for
consistency's sake
(closes issue #11381, reported and patched by jon)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89621
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Mark Michelson [Mon, 26 Nov 2007 23:11:29 +0000 (23:11 +0000)]
Merged revisions 89618 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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Mark Michelson [Mon, 26 Nov 2007 23:05:12 +0000 (23:05 +0000)]
Blocked revisions 89616 via svnmerge
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r89616 | mmichelson | 2007-11-26 17:02:30 -0600 (Mon, 26 Nov 2007) | 5 lines
After issuing a "say load new" tons of warning messages are printed
out to the CLI every time do_say in app_playback is called. Removing these
warnings
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Russell Bryant [Mon, 26 Nov 2007 22:52:36 +0000 (22:52 +0000)]
Update the configure script check for libpri to check for the newest function
that was just added.
Cresl1n, please keep this in mind when making these changes to libpri or libss7.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89615
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Olle Johansson [Mon, 26 Nov 2007 21:23:48 +0000 (21:23 +0000)]
Rename "limitonpeer" to "counteronpeer" since the call-limit is deprecated.
Both still works in this version.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89613
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Joshua Colp [Mon, 26 Nov 2007 21:14:07 +0000 (21:14 +0000)]
Merged revisions 89610 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89610 | file | 2007-11-26 17:10:29 -0400 (Mon, 26 Nov 2007) | 2 lines
Fix issues with async dialing with an application executing. The application has to be terminated and control returned to the thread before hanging things up. (issue #BE-252)
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Olle Johansson [Mon, 26 Nov 2007 21:12:50 +0000 (21:12 +0000)]
Formatting, doxygenification
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89611
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Olle Johansson [Mon, 26 Nov 2007 20:55:09 +0000 (20:55 +0000)]
Formatting changes, cleaning up some code
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Olle Johansson [Mon, 26 Nov 2007 20:19:50 +0000 (20:19 +0000)]
Start using Doxygen groupings to group variables and defines.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89607
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Olle Johansson [Mon, 26 Nov 2007 19:24:23 +0000 (19:24 +0000)]
- Mark "concise" as deprecated
- 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
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Joshua Colp [Mon, 26 Nov 2007 18:11:31 +0000 (18:11 +0000)]
Perform some module use counting audits. This is now done outside the scope of the application/dialplan function so they do not need to worry about it.
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Joshua Colp [Mon, 26 Nov 2007 18:04:46 +0000 (18:04 +0000)]
Merged revisions 89599 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89599 | file | 2007-11-26 14:02:56 -0400 (Mon, 26 Nov 2007) | 6 lines
Add module counting removal for error conditions.
(closes issue #11333)
Reported by: Laureano
Patches:
res_features_v2.c.patch uploaded by Laureano (license 265)
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Russell Bryant [Mon, 26 Nov 2007 17:49:47 +0000 (17:49 +0000)]
Merged revisions 89594 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89594 | russell | 2007-11-26 11:41:04 -0600 (Mon, 26 Nov 2007) | 3 lines
Add channel locking to a function that needed to be doing it. This is just a
little something I noticed while working on a completely unrelated issue.
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Steve Murphy [Mon, 26 Nov 2007 17:46:41 +0000 (17:46 +0000)]
closes issue #11341; made changes to make utils again right with the MTX_PROFILE world.
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