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r132390 | russell | 2008-07-21 09:47:41 -0500 (Mon, 21 Jul 2008) | 16 lines
Remove libresample from the Asterisk source tree. It is now available in its
own repository, and must be installed like any other library for Asterisk to
use. The two modules that require it are codec_resample and app_jack.
To install libresample:
$ svn co http://svn.digium.com/svn/libresample/trunk libresample
$ cd libresample
$ ./configure
$ make
$ sudo make install
This code is currently in our own repository because the build system did not
include the appropriate targets for building a dynamic library or for installing
the library.
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r132242 | bbryant | 2008-07-18 17:19:56 -0500 (Fri, 18 Jul 2008) | 4 lines
Fixes problem where manager users loaded from users.conf would be
removed early (before the routine to load the configuration was
finished) because a variable wasn't initialized.
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r132206 | tilghman | 2008-07-18 15:57:47 -0500 (Fri, 18 Jul 2008) | 5 lines
Russell pointed out that using ast_strdupa() within a loop like this is
probably not a good idea, as we might run out of stack space. Therefore,
changing this over to use the ast_str infrastructure for buffers is
probably a good idea.
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r132050 | bbryant | 2008-07-18 12:55:41 -0500 (Fri, 18 Jul 2008) | 8 lines
Fix magic Revision keywords in hashtab.c and change cdr_radius.c to use
the same keyword as the other files (patch by eliel).
(closes issue #13104)
Reported by: eliel
Patches:
revision.patch uploaded by eliel (license 64)
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r131868 | jpeeler | 2008-07-17 17:40:00 -0500 (Thu, 17 Jul 2008) | 6 lines
Add configuration option to chan_dahdi.conf to allow buffering policy and number of buffers to be configured per channel. Syntax:
buffers=<num of buffers>,<policy>
Where the number of buffers is some non-negative integer and the policy is either "full", "half", or "immediate".
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r131824 | mmichelson | 2008-07-17 16:26:41 -0500 (Thu, 17 Jul 2008) | 10 lines
Document that the duration of dtmf may be passed to
the SendDTMF application. Also correct the default
pause between digits.
(closes issue #13102)
Reported by: eliel
Patches:
app_senddtmf.c.patch uploaded by eliel (license 64)
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r131717 | bbryant | 2008-07-17 13:14:42 -0500 (Thu, 17 Jul 2008) | 8 lines
Fix a memory leak in register_group_feature when attempting to register
a feature without specifying a group or feature to register.
(closes issue #13101)
Reported by: eliel
Patches:
features.c.patch uploaded by eliel (license 64)
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r131643 | russell | 2008-07-17 09:46:29 -0500 (Thu, 17 Jul 2008) | 5 lines
Instead of attempting to pass through AST_EVENT_DEVICE_STATE, use DEVICE_STATE_CHANGE
instead. DEVICE_STATE is a state change on one server, and DEVICE_STATE_CHANGE is
the "real" state of that device across all servers sharing state. This would have
only been a problem with distributed device state.
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as it is not as necessary, because log_show_lock in trunk
is not available in 1.6.0, and is estimated to be the
only function that might care about the lock_addr values.
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r131445 | bbryant | 2008-07-16 16:24:18 -0500 (Wed, 16 Jul 2008) | 9 lines
Fixes an issue with "core show sysinfo" that used the wrong operator to
calculate the number of bytes from a sysinfo structure.
unsigned long.
(closes issue #13057)
Reported by: eliel
Patches:
asterisk.c.patch uploaded by eliel (license 64)
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r131375 | mmichelson | 2008-07-16 15:24:12 -0500 (Wed, 16 Jul 2008) | 22 lines
Merged revisions 131369 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r131369 | mmichelson | 2008-07-16 15:23:02 -0500 (Wed, 16 Jul 2008) | 14 lines
Move the init_queue call back to where it used to be (changed
Sept 12 last year). It was moved then to prevent a memory leak.
Since then, the same memory leak recurred and was fixed in a
better way.
Now it has been found that the placement of this init_queue
call can cause problems if a realtime queue has values changed
to an empty string. The problem is that the default value
for that queue parameter would not be set.
(closes issue #13084)
Reported by: elbriga
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r131300 | mmichelson | 2008-07-16 13:59:27 -0500 (Wed, 16 Jul 2008) | 21 lines
Merged revisions 131299 via svnmerge from
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r131299 | mmichelson | 2008-07-16 13:57:34 -0500 (Wed, 16 Jul 2008) | 13 lines
Make absolutely certain that the transfer datastore
is removed from the calling channel once the caller
is finished in the queue. This could have weird con-
sequences when dialing local queue members when multiple
transfers occur on a single call.
Also fixed a memory leak that would occur when an
attended transfer occurred from a queue member.
(closes issue #13047)
Reported by: festr
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r131243 | murf | 2008-07-16 11:59:33 -0600 (Wed, 16 Jul 2008) | 27 lines
Merged revisions 131242 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r131242 | murf | 2008-07-16 11:53:43 -0600 (Wed, 16 Jul 2008) | 19 lines
(closes issue #13090)
Reported by: murf
The problem was that, esoteric as it is, because the hangerupper
context immediately preceded the std-priv-extent macro, that
the checking code accidentally would fall from traversing hangerupper
into the std-priv-exten macro, where it would hit the hangerupper
in the 'includes', and proceed into an infinite recursion.
A small fix to traverse into the statements of the context instead
of the context solves this issue.
I also added some commented out printfs for debug, which were pretty
handy in the face of a dorky gdb.
This was a problem around since the package was first written;
but evidently pretty rare in turning up in the field.
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r131129 | murf | 2008-07-15 17:36:19 -0600 (Tue, 15 Jul 2008) | 21 lines
(closes issue #12960)
Reported by: mnicholson
Spent most of the day on this bug, and the
solution was so simple. Just had to find and
understand the problem.
The problem was, that the routine to copy
the existing switches, includes, and ignorepats
from the old context to the new one, wasn't
getting called when the context is already
existent. (In other words, if AEL is adding
a new context to the mix, they get copied,
but if pbx_config already defined a context,
then the copy wasn't happening. This made
no sense, so I moved the call to copy the
includes & etc, no matter the case.
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r131072 | russell | 2008-07-15 13:46:40 -0500 (Tue, 15 Jul 2008) | 5 lines
Fix a couple of places in res_agi where the agi_commands lock would not be
released, causing a deadlock. (Reported by mvanbaak in #asterisk-dev,
discovered by bbryant's change to the lock tracking code to yell at you
if a thread exits with a lock still held)
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Merging this rev from trunk to 1.6.0 was not
simple. Why? Because we've enhanced trunk to
do a [fast] merge-and-delete operation which
also solved problems with contexts having
entries from different registrars.
Fast as in the amount of time the contexts
are locked down. That *is* fast, but traversing
the entire dialplan looking for priorities to
delete takes more time overall.
This particular fix involved pulling in those
enhancements from trunk, along with all the
various fixes and refinements made along the
way.
Merging all this from trunk into 1.6 involved:
a. mergetrunk6 in the stuff from 130145;
b. revert all but the prop changes
c. catalog all revisions to pbx.c since 1.6.0 was forked
(at rev 105596).
d. catalog all revisions to pbx.c in trunk since 1.6.0
was forked, making special note of all revs that
were not merged into 1.6.0.
e. study each rev in trunk not applied to 1.6.0, and
determine if it was involved in the merge_and_delete
enhancements in trunk. 25 commits were done in 1.6.0,
all but one (106306) was a merge from trunk.
Trunk had 22 additional changes, of which 7 were
involved in the merge_and_delete enhancements:
106757
108894
109169
116461
123358
130145
130297
f. Go to trunk and collect patches, one by one,
of the changes made by each rev across the
entire source tree, using svn diff -c <num> > pfile
g. Apply each patch in order to 1.6.0, and
resolve all failures and compilation problems
before proceding to the next patch.
h. test the stuff.
i. profit!
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r130145 | murf | 2008-07-11 12:24:31 -0600 (Fri, 11 Jul 2008) | 40 lines
(closes issue #13041)
Reported by: eliel
Tested by: murf
(closes issue #12960)
Reported by: mnicholson
In this 'omnibus' fix, I **think** I solved both
the problem in 13041, where unloading pbx_ael.so
caused crashes, or incomplete removal of previous
registrar'ed entries. And I added code to completely
remove all includes, switches, and ignorepats that
had a matching registrar entry, which should
appease 12960.
I also added a lot of seemingly useless brackets
around single statement if's, which helped debug
so much that I'm leaving them there.
I added a routine to check the correlation between
the extension tree lists and the hashtab
tables. It can be amazingly helpful when you have
lots of dialplan stuff, and need to narrow
down where a problem is occurring. It's ifdef'd
out by default.
I cleaned up the code around the new CIDmatch code.
It was leaving hanging extens with bad ptrs, getting confused
over which objects to remove, etc. I tightened
up the code and changed the call to remove_exten
in the merge_and_delete code.
I added more conditions to check for empty context
worthy of deletion. It's not empty if there are
any includes, switches, or ignorepats present.
If I've missed anything, please re-open this bug,
and be prepared to supply example dialplan code.
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r130890 | tilghman | 2008-07-14 18:59:54 -0500 (Mon, 14 Jul 2008) | 16 lines
Merged revisions 130889 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r130889 | tilghman | 2008-07-14 18:59:13 -0500 (Mon, 14 Jul 2008) | 8 lines
Override the callerid in all cases when the callerid is set in the user, not
just when a remote callerid is set. Also, if not set in the user, allow the
remote CallerID to pass through.
(closes issue #12875)
Reported by: dimas
Patches:
20080714__bug12875.diff.txt uploaded by Corydon76 (license 14)
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r130794 | mmichelson | 2008-07-14 12:54:11 -0500 (Mon, 14 Jul 2008) | 16 lines
Merged revisions 130792 via svnmerge from
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r130792 | mmichelson | 2008-07-14 12:50:21 -0500 (Mon, 14 Jul 2008) | 8 lines
Add a check to the CAN_EARLY_BRIDGE macro in app_dial to
be sure there are no audiohooks present on the channels
involved. This fixed a one-way audio situation I had in
my test setup. I couldn't find any open issues that suggested
one-way audio with regards to mixmonitor (or other audiohook)
usage, though.
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