2010-06-01 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.9-rc1 Released.
2010-06-01 15:20 +0000 [r266598] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c, /: Merged revisions 266592 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r266592 | tilghman | 2010-06-01 10:18:59 -0500 (Tue, 01 Jun 2010)
| 18 lines Merged revisions 266585 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r266585 | tilghman | 2010-06-01 10:17:46 -0500 (Tue, 01 Jun 2010)
| 11 lines Prevent CLI prompt from distorting output of lines
shorter than the prompt. Uses the VT100 method of clearing the
line from the cursor position to the end of the line: Esc-0K
(closes issue #17160) Reported by: coolmig Patches:
20100531__issue17160.diff.txt uploaded by tilghman (license 14)
Tested by: coolmig ........ ................
2010-05-31 16:07 +0000 [r266570] Paul Belanger <paul.belanger@polybeacon.com>
* res/res_agi.c: Fix typo in documentation (closes issue #17395)
Reported by: pabelanger Patches: res_agi.c.patch uploaded by
pabelanger (license 224)
2010-05-30 04:45 +0000 [r266439] Tilghman Lesher <tlesher@digium.com>
* contrib/init.d/rc.debian.asterisk, /: Merged revisions 266438 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r266438 | tilghman | 2010-05-29 23:44:28 -0500
(Sat, 29 May 2010) | 9 lines Merged revisions 266437 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r266437 | tilghman | 2010-05-29 23:43:28 -0500 (Sat, 29
May 2010) | 2 lines Reverting patch and reopening issue #16784,
as patch breaks color display. ........ ................
2010-05-28 20:55 +0000 [r266338] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c, /: Merged revisions 266337 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r266337 |
tilghman | 2010-05-28 15:53:04 -0500 (Fri, 28 May 2010) | 1 line
Only report swap on platforms which can examine those statistics
........
2010-05-28 17:57 +0000 [r266293] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 266292 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r266292 |
dvossel | 2010-05-28 12:55:38 -0500 (Fri, 28 May 2010) | 9 lines
fixes crash when creation of UDPTL fails (closes issue #17264)
Reported by: falves11 Patches: issue_17264_reviewboard_fix.diff
uploaded by dvossel (license 671)
issue_17264_1.6.2_reviewboard_fix.diff uploaded by dvossel
(license 671) Tested by: falves11 ........
2010-05-26 21:19 +0000 [r266154] Tilghman Lesher <tlesher@digium.com>
* utils/extconf.c, main/asterisk.c, /, main/logger.c: Merged
revisions 266146 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r266146 | tilghman | 2010-05-26 16:17:46 -0500 (Wed, 26 May 2010)
| 21 lines Merged revisions 266142 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r266142 | tilghman | 2010-05-26 16:11:44 -0500 (Wed, 26 May 2010)
| 14 lines Use sigaction for signals which should persist past
the initial trigger, not signal. If you call signal() in a
Solaris signal handler, instead of just resetting the signal
handler, it causes the signal to refire, because the signal is
not marked as handled prior to the signal handler being called.
This effectively causes Solaris to immediately exceed the
threadstack in recursive signal handlers and crash. (closes issue
#17000) Reported by: rmcgilvr Patches:
20100526__issue17000.diff.txt uploaded by tilghman (license 14)
Tested by: rmcgilvr ........ ................
2010-05-26 18:37 +0000 [r266007] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 266006 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r266006 |
dvossel | 2010-05-26 13:32:51 -0500 (Wed, 26 May 2010) | 8 lines
fixes failed SIP Directed pickup resulting in dead channel
(closes issue #17339) Reported by: one47 Patches:
sip_magic_pickup2 uploaded by one47 (license 23) Tested by:
one47, dvossel ........
2010-05-26 16:31 +0000 [r265895-265959] Tilghman Lesher <tlesher@digium.com>
* res/res_config_pgsql.c, /: Merged revisions 265923 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r265923 | tilghman | 2010-05-26 11:23:28 -0500
(Wed, 26 May 2010) | 14 lines Merged revisions 265910 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r265910 | tilghman | 2010-05-26 11:21:00 -0500 (Wed, 26 May 2010)
| 7 lines Not finding rows in the DB does not rise to the level
of a warning. (closes issue #17062) Reported by: drookie Patches:
20100525__issue17062.diff.txt uploaded by tilghman (license 14)
........ ................
* configs/res_pgsql.conf.sample, res/res_config_pgsql.c, /: Merged
revisions 265894 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r265894 |
tilghman | 2010-05-26 11:14:48 -0500 (Wed, 26 May 2010) | 8 lines
Construct socket name, according to the Postgres docs, and
document as such. (closes issue #17392) Reported by: dps Patches:
20100525__issue17392.diff.txt uploaded by tilghman (license 14)
Tested by: dps ........
2010-05-26 15:52 +0000 [r265890] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Recorded merge of revisions 265842 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r265842 | mmichelson | 2010-05-26 09:41:55 -0500 (Wed,
26 May 2010) | 9 lines Re-enable "always" option for videosupport
option in sip.conf. (closes issue #17016) Reported by: twilson
Patches: 17016.patch uploaded by mmichelson (license 60) Tested
by: devmod ........
2010-05-26 00:33 +0000 [r265748] Tilghman Lesher <tlesher@digium.com>
* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
pbx/pbx_lua.c: Merged revisions 265747 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r265747 |
tilghman | 2010-05-25 19:29:40 -0500 (Tue, 25 May 2010) | 8 lines
Use configure to determine the prefixes and include directories
properly. This ensures cross-platform compatibility, even among
Linux distributions, which don't always put headers in the same
place. (closes issue #17391) Reported by: loloski ........
2010-05-25 21:05 +0000 [r265699] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 265698 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r265698 |
mmichelson | 2010-05-25 15:59:04 -0500 (Tue, 25 May 2010) | 12
lines Properly use peer's outboundproxy for outbound REGISTERs.
The logic used in transmit_register to get the outboundproxy for
a peer was flawed since this value would be overridden shortly
afterwards when create_addr was called. In addition, this also
fixes some logic used when parsing users.conf so that the peer
name is placed in the internally-generated register string so
that an outboundproxy set in the Asterisk GUI will be used for
outbound REGISTERs. ........
2010-05-25 17:15 +0000 [r265615] David Vossel <dvossel@digium.com>
* channels/chan_dahdi.c: fixes build issue with zaptel (closes
issue 0017394) Reported by: aragon Patches: half_buffer_fix.diff
uploaded by dvossel (license 671) Tested by: aragon
2010-05-25 17:06 +0000 [r265612] Matthew Nicholson <mnicholson@digium.com>
* apps/app_queue.c, /: Merged revisions 265611 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r265611 | mnicholson | 2010-05-25 12:00:11 -0500 (Tue, 25 May
2010) | 15 lines Merged revisions 265610 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r265610 | mnicholson | 2010-05-25 11:48:19 -0500 (Tue, 25 May
2010) | 8 lines Don't mark the cdr records of unanswered queue
calls with "NOANSWER". This restores the behavior prior to
r258670. (closes issue #17334) Reported by: jvandal Patches:
queue-cdr-fixes1.diff uploaded by mnicholson (license 96) Tested
by: aragon, jvandal ........ ................
2010-05-24 23:52 +0000 [r265521] Terry Wilson <twilson@digium.com>
* include/asterisk/options.h, main/asterisk.c, Makefile,
doc/manager_1_1.txt, doc/tex/manager.tex, main/manager.c: Merged
revisions 265320,265467 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r265320 |
twilson | 2010-05-24 14:06:40 -0500 (Mon, 24 May 2010) | 14 lines
Add the FullyBooted AMI event It is possible to connect to the
manager interface before all Asterisk modules are loaded. To
ensure that an application does not send AMI actions that might
require a module that has not yet loaded, the application can
listen for the FullyBooted manager event. It will be sent upon
connection if all modules have been loaded, or as soon as loading
is complete. The event: Event: FullyBooted Privilege: system,all
Status: Fully Booted Review:
https://reviewboard.asterisk.org/r/639/ ........ r265467 |
twilson | 2010-05-24 17:21:58 -0500 (Mon, 24 May 2010) | 1 line
Merge the rest of the FullyBooted patch ........
2010-05-24 22:07 +0000 [r265450-265452] Mark Michelson <mmichelson@digium.com>
* /, channels/h323/ast_h323.cxx: Merged revisions 265451 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r265451 | mmichelson | 2010-05-24 17:05:15 -0500 (Mon,
24 May 2010) | 8 lines Print openh323 log to the Asterisk
console. (closes issue #17109) Reported by: under Patches:
logstream.diff uploaded by under (license 914) ........
* /, channels/chan_sip.c: Merged revisions 265449 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r265449 |
mmichelson | 2010-05-24 16:44:30 -0500 (Mon, 24 May 2010) | 11
lines Allow type=user SIP endpoints to be loaded properly from
realtime. (closes issue #16021) Reported by: Guggemand Patches:
realtime-type-fix.patch uploaded by Guggemand (license 897)
(altered by me slightly to avoid ref leaks) Tested by: Guggemand
........
2010-05-24 19:30 +0000 [r265364] David Vossel <dvossel@digium.com>
* main/channel.c, /: Merged revisions 265273 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r265273 |
dvossel | 2010-05-24 11:10:09 -0500 (Mon, 24 May 2010) | 2 lines
fixes segfault when using generic plc ........
2010-05-24 18:30 +0000 [r265318] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c, /: Merged revisions 265316 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r265316 |
tilghman | 2010-05-24 13:19:08 -0500 (Mon, 24 May 2010) | 7 lines
On systems with a LOT of RAM, a signed integer sometimes printed
negative. (closes issue #16837) Reported by: jlpedrosa Patches:
20100504__issue16837.diff.txt uploaded by tilghman (license 14)
........
2010-05-21 21:57 +0000 [r264998-265172] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fix memory hogging behavior of app_queue. From
reviewboard: This review request is for the patch on issue 17081.
A user reported that he saw increasing numbers of allocations
stemming from app_queue.c when he would run the "queue show" CLI
command. The user reported that he was using approximately 40
realtime queues and as he ran the CLI command more and more, the
memory usage would shoot up. As it turns out, there was a memory
leak and a separate usage of memory that, while not really a
leak, was very irresponsible. Both memory problems can be
attributed to the function init_queue(). When the "queue show"
command is run, all realtime queues have the init_queue()
function called on the in-memory queue. The idea is to place the
queue in its default state and then overwrite options specified
in the realtime backend as we read them. The first problem, the
memory leak, had to do with the fact that the string field for
the name of the first periodic announcement file was being
re-created every time init_queue was called. This patch corrects
the behavior by only calling ast_str_create if the memory has not
already been allocated. The other problem is a bit more
complicated. The majority of the strings in the call_queue
structure were changed to use the ast_string_fields API for 1.6.0
and beyond. init_queue resets all string fields on the queue to
their default values. Then, later in the realtime queue loading
process, these string fields are set to their configured values.
For those unfamiliar with string fields, frequent resizing of a
string like this is not what the string fields API is designed
for. The result of this constant resizing is that as the queue
gets loaded, eventually space for the string runs out and so a
new memory pool, at twice the size of the previously allocated
one, is created for the string fields. The reporter of issue
17081 wrote a script that ran the "queue show" CLI command 2100
times. By the end, each of his 40 queues was taking about a
megabyte of memory apiece just for their string fields. My fix
for this problem is to revert the call_queue structure from using
string fields. In my patch here, I have moved the queue back to
using fixed-sized buffers. I ran the script provided by the
reporter of 17081 and determined that I no longer saw the
steadily-increasing memory usage that I had seen before applying
the patch. (closes issue #17081) Reported by: wliegel Patches:
17081v2.patch uploaded by mmichelson (license 60) Tested by:
wliegel, mmichelson Review:
https://reviewboard.asterisk.org/r/651/
* apps/app_queue.c, include/asterisk/file.h, /: Merged revisions
265090 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r265090 | mmichelson | 2010-05-21 16:08:51 -0500 (Fri, 21 May
2010) | 15 lines Merged revisions 265089 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r265089 | mmichelson | 2010-05-21 15:59:14 -0500 (Fri, 21 May
2010) | 8 lines Don't hang up on a queue caller if the file we
attempt to play does not exist. This also fixes a documentation
mistake in file.h that made my original attempt to correct this
problem not work correctly. (closes issue #17061) Reported by:
RoadKill ........ ................
* /, channels/chan_sip.c: Merged revisions 265087 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r265087 |
mmichelson | 2010-05-21 15:38:14 -0500 (Fri, 21 May 2010) | 7
lines Be sure to set the sin_family on the proxy when allocating.
(closes issue #17157) Reported by: stuarth ........
* /, include/asterisk/channel.h: Merged revisions 265000 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r265000 | mmichelson | 2010-05-21 11:54:21 -0500
(Fri, 21 May 2010) | 9 lines Merged revisions 264999 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r264999 | mmichelson | 2010-05-21 11:53:53 -0500 (Fri,
21 May 2010) | 3 lines Fix grammatical error in comment. ........
................
* main/channel.c, main/autoservice.c, /,
include/asterisk/channel.h: Merged revisions 264997 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r264997 | mmichelson | 2010-05-21 11:44:27 -0500
(Fri, 21 May 2010) | 38 lines Merged revisions 264996 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r264996 | mmichelson | 2010-05-21 11:28:34 -0500 (Fri, 21 May
2010) | 32 lines Allow ast_safe_sleep to defer specific frames
until after the sleep has concluded. From reviewboard Background:
A Digium customer discovered a somewhat odd bug. The setup is
that parties A and B are bridged, and party A places party B on
hold. While party B is listening to hold music, he mashes a bunch
of DTMF. Party A takes party B off hold while this is happening,
but party B continues to hear hold music. I could reproduce this
about 1 in 5 times. The issue: When DTMF features are enabled and
a user presses keys, the channel that the DTMF is streamed to is
placed in an ast_safe_sleep for 100 ms, the duration of the
emulated tone. If an AST_CONTROL_UNHOLD frame is read from the
channel during the sleep, the frame is dropped. Thus the unhold
indication is never made to the channel that was originally
placed on hold. The fix: Originally, I discussed with Kevin
possible ways of fixing the specific problem reported. However,
we determined that the same type of problem could happen in other
situations where ast_safe_sleep() is used. Using autoservice as a
model, I modified ast_safe_sleep_conditional() to defer specific
frame types so they can be re-queued once the sleep has finished.
I made a common function for determining if a frame should be
deferred so that there are not two identical switch blocks to
maintain. Review: https://reviewboard.asterisk.org/r/674/
........ ................
2010-05-20 23:34 +0000 [r264829] Richard Mudgett <rmudgett@digium.com>
* /, main/callerid.c: Merged revisions 264828 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r264828 | rmudgett | 2010-05-20 18:29:43 -0500 (Thu, 20 May 2010)
| 13 lines Merged revisions 264820 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r264820 | rmudgett | 2010-05-20 18:23:21 -0500 (Thu, 20 May 2010)
| 6 lines ast_callerid_parse() had a path that left name
uninitialized. Several callers of ast_callerid_parse() do not
initialize the name parameter before calling thus there is the
potential to use an uninitialized pointer. ........
................
2010-05-20 22:24 +0000 [r264753-264783] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c, /: Merged revisions 264779 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r264779 |
tilghman | 2010-05-20 17:23:32 -0500 (Thu, 20 May 2010) | 8 lines
Let ExtensionState resolve dynamic hints. (closes issue #16623)
Reported by: tilghman Patches: 20100116__issue16623.diff.txt
uploaded by tilghman (license 14) Tested by: lmadsen ........
* apps/app_stack.c, /: Merged revisions 264752 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r264752 |
tilghman | 2010-05-20 16:28:53 -0500 (Thu, 20 May 2010) | 7 lines
Error message fix. (closes issue #17356) Reported by: kenner
Patches: app_stack.c.diff uploaded by kenner (license 1040)
........
2010-05-19 22:10 +0000 [r264453] Mark Michelson <mmichelson@digium.com>
* include/asterisk/_private.h, include/asterisk/options.h,
main/asterisk.c, main/loader.c, main/channel.c, /,
channels/chan_sip.c: Merged revisions 264452 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r264452 |
mmichelson | 2010-05-19 16:29:08 -0500 (Wed, 19 May 2010) | 86
lines Fix transcode_via_sln option with SIP calls and improve PLC
usage. From reviewboard: The problem here is a bit complex, so
try to bear with me... It was noticed by a Digium customer that
generic PLC (as configured in codecs.conf) did not appear to
actually be having any sort of benefit when packet loss was
introduced on an RTP stream. I reproduced this issue myself by
streaming a file across an RTP stream and dropping approx. 5% of
the RTP packets. I saw no real difference between when PLC was
enabled or disabled when using wireshark to analyze the RTP
streams. After analyzing what was going on, it became clear that
one of the problems faced was that when running my tests, the
translation paths were being set up in such a way that PLC could
not possibly work as expected. To illustrate, if packets are lost
on channel A's read stream, then we expect that PLC will be
applied to channel B's write stream. The problem is that generic
PLC can only be done when there is a translation path that moves
from some codec to SLINEAR. When I would run my tests, I found
that every single time, read and write translation paths would be
set up on channel A instead of channel B. There appeared to be no
real way to predict which channel the translation paths would be
set up on. This is where Kevin swooped in to let me know about
the transcode_via_sln option in asterisk.conf. It is supposed to
work by placing a read translation path on both channels from the
channel's rawreadformat to SLINEAR. It also will place a write
translation path on both channels from SLINEAR to the channel's
rawwriteformat. Using this option allows one to predictably set
up translation paths on all channels. There are two problems with
this, though. First and foremost, the transcode_via_sln option
did not appear to be working properly when I was placing a SIP
call between two endpoints which did not share any common
formats. Second, even if this option were to work, for PLC to be
applied, there had to be a write translation path that would go
from some format to SLINEAR. It would not work properly if the
starting format of translation was SLINEAR. The one-line change
presented in this review request in chan_sip.c fixed the first
issue for me. The problem was that in sip_request_call, the
jointcapability of the outbound channel was being set to the
format passed to sip_request_call. This is nativeformats of the
inbound channel. Because of this, when
ast_channel_make_compatible was called by app_dial, both channels
already had compatibly read and write formats. Thus, no
translation path was set up at the time. My change is to set the
jointcapability of the sip_pvt created during sip_request_call to
the intersection of the inbound channel's nativeformats and the
configured peer capability that we determined during the earlier
call to create_addr. Doing this got the translation paths set up
as expected when using transcode_via_sln. The changes presented
in channel.c fixed the second issue for me. First and foremost,
when Asterisk is started, we'll read codecs.conf to see the value
of the genericplc option. If this option is set, and ast_write is
called for a frame with no data, then we will attempt to fill in
the missing samples for the frame. The implementation uses a
channel datastore for maintaining the PLC state and for creating
a buffer to store PLC samples in. Even when we receive a frame
with data, we'll call plc_rx so that the PLC state will have
knowledge of the previous voice frame, which it can use as a
basis for when it comes time to actually do a PLC fill-in. So,
reviewers, now I ask for your help. First off, there's the one
line change in chan_sip that I have put in. Is it right? By my
logic it seems correct, but I'm sure someone can tell me why it
is not going to work. This is probably the change I'm least
concerned about, though. What concerns me much more is the set of
changes in channel.c. First off, am I even doing it right? When I
run tests, I can clearly see that when PLC is activated, I see a
significant increase in RTP traffic where I would expect it to
be. However, in my humble opinion, the audio sounds kind of
crappy whenever the PLC fill-in is done. It sounds worse to me
than when no PLC is used at all. I need someone to review the
logic I have used to be sure that I'm not misusing anything. As
far as I can see my pointer arithmetic is correct, and my use of
AST_FRIENDLY_OFFSET should be correct as well, but I'm sure
someone can point out somewhere where I've done something
incorrectly. As I was writing this review request up, I decided
to give the code a test run under valgrind, and I find that for
some reason, calls to plc_rx are causing some invalid reads.
Apparently I'm reading past the end of a buffer somehow. I'll
have to dig around a bit to see why that is the case. If it's
obvious to someone reviewing, speak up! Finally, I have one other
proposal that is not reflected in my code review. Since without
transcode_via_sln set, one cannot predict or control where a
translation path will be up, it seems to me that the current
practice of using PLC only when transcoding to SLINEAR is not
useful. I recommend that once it has been determined that the
method used in this code review is correct and works as expected,
then the code in translate.c that invokes PLC should be removed.
Review: https://reviewboard.asterisk.org/r/622/ ........
2010-05-19 20:31 +0000 [r264405] David Vossel <dvossel@digium.com>
* main/udptl.c, /: Merged revisions 264400 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r264400 |
dvossel | 2010-05-19 15:30:33 -0500 (Wed, 19 May 2010) | 11 lines
fixes infinite loop during udptl.c's decode_open_type When
decode_length returns the length there is a check to see if that
length is negative, if so the decode loop breaks as this means
the limit has been reached. The problem here is that length is an
unsigned int, so length can never be negative. This resulted in
an infinite loop. (issue #17352) ........
2010-05-19 20:27 +0000 [r264336-264388] Matthew Nicholson <mnicholson@digium.com>
* main/udptl.c, /: Merged revisions 264379 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r264379 |
mnicholson | 2010-05-19 15:26:27 -0500 (Wed, 19 May 2010) | 4
lines Cast an unsigned int to a signed int when comparing it with
0. (AST-377) ........
* apps/app_speech_utils.c, /: Merged revisions 264335 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r264335 | mnicholson | 2010-05-19 15:02:57 -0500
(Wed, 19 May 2010) | 12 lines Merged revisions 264334 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r264334 | mnicholson | 2010-05-19 15:01:38 -0500 (Wed, 19 May
2010) | 5 lines Set quieted flag when receiving a dtmf tone
during playback in speechbackground. (closes issue #16966)
Reported by: asackheim ........ ................
2010-05-19 19:25 +0000 [r264332] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 264331 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r264331 |
dvossel | 2010-05-19 14:21:04 -0500 (Wed, 19 May 2010) | 13 lines
fixes crash in check_rtp_timeout During deadlock avoidance the
sip dialog pvt is locked and unlocked. When this occurs we have
no guarantee the pvt's owner is still valid. We were trying to
access the pvt's owner after this without checking to see if it
still existed first. (closes issue #17271) Reported by: under
Patches: check_rtp_timeout.diff uploaded by under (license 914)
Tested by: dvossel ........
2010-05-19 17:49 +0000 [r264205-264250] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/options.h, /, configure,
include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
264249 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r264249 | tilghman | 2010-05-19 12:48:31 -0500 (Wed, 19 May 2010)
| 24 lines Merged revisions 264248 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r264248 | tilghman | 2010-05-19 12:41:29 -0500 (Wed, 19 May 2010)
| 17 lines Internal timing is now on by default, if you're using
DAHDI 2.3 or above. The reason for ensuring DAHDI 2.3 or above is
that this version ensures that a timer is always available,
whereas in previous versions, it was possible for DAHDI to be
loaded, but have no drivers to actually generate timing. If
internal_timing was turned on in this circumstance, a complete
lack of audio would result. This is the reason why
internal_timing was not on by default. However, now that DAHDI
ensures the availability of a timer, there is no reason for this
setting to be off (and in fact, it solves a great many initial
user problems). (closes issue #15932) Reported by: dimas Patches:
20100519__issue15932.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman ........ ................
* main/dsp.c, /: Merged revisions 264204 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r264204 |
tilghman | 2010-05-19 11:42:20 -0500 (Wed, 19 May 2010) | 9 lines
Keep track of digit duration, when we're decoding inband to pass
DTMF frames. (closes issue #17235) Reported by: frawd Patches:
new_dtmf_dsp_len.patch uploaded by frawd (license 610)
20100518__issue17235.diff.txt uploaded by tilghman (license 14)
Tested by: frawd ........
2010-05-19 14:47 +0000 [r264115] David Vossel <dvossel@digium.com>
* main/rtp.c, /: Merged revisions 264114 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r264114 |
dvossel | 2010-05-19 09:38:02 -0500 (Wed, 19 May 2010) | 13 lines
fixes crash during dtmf During the processing of Cisco dtmf the
dtmf samples were not being calculated correctly. In an attempt
to determine what sample rate was being used, a NULL frame was
processed which caused a crash. This patch resolves this. (closes
issue #17248) Reported by: falves11 Patches: issue_17248.diff
uploaded by dvossel (license 671) ........
2010-05-19 08:15 +0000 [r264032] Alec L Davis <sivad.a@paradise.net.nz>
* /, configs/indications.conf.sample: Merged revisions 264031 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r264031 | alecdavis | 2010-05-19 20:09:14 +1200 (Wed, 19
May 2010) | 8 lines fix incorrectly typed indications for [nz]
stutter and dialrecall (closes issue #17359) Reported by:
alecdavis Patches: bug17359.diff.txt uploaded by alecdavis
(license 585) ........
2010-05-19 06:41 +0000 [r263951] Tilghman Lesher <tlesher@digium.com>
* main/dsp.c, /: Merged revisions 263950 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r263950 | tilghman | 2010-05-19 01:41:04 -0500 (Wed, 19 May 2010)
| 15 lines Merged revisions 263949 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r263949 | tilghman | 2010-05-19 01:32:27 -0500 (Wed, 19 May 2010)
| 8 lines Because progress is called multiple times, across
several frames, we must persist states when detecting multitone
sequences. (closes issue #16749) Reported by: dant Patches:
dsp.c-bug16749-1.patch uploaded by dant (license 670) Tested by:
dant ........ ................
2010-05-18 22:49 +0000 [r263906] David Vossel <dvossel@digium.com>
* main/strings.c, /: Merged revisions 263904 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r263904 |
dvossel | 2010-05-18 17:48:51 -0500 (Tue, 18 May 2010) | 9 lines
fixes segfault on logging (closes issue #17331) Reported by:
under Patches: utils.diff uploaded by under (license 914)
segfault_on_logging.diff uploaded by dvossel (license 671) Tested
by: under, dvossel ........
2010-05-18 19:41 +0000 [r263809] Jeff Peeler <jpeeler@digium.com>
* apps/app_directory.c, /: Merged revisions 263807 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r263807 | jpeeler | 2010-05-18 14:27:34 -0500
(Tue, 18 May 2010) | 17 lines Merged revisions 263769 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r263769 | jpeeler | 2010-05-18 13:54:58 -0500 (Tue, 18 May 2010)
| 10 lines Modify directory name reading to be interrupted with
operator or pound escape. In the case of accidentally entering
the wrong first three letters for the reading, users could be
very frustrated if the name listing is very long. This allows
interrupting the reading by pressing 0 or #. 0 will attempt to
execute a configured operator (o) extension and # will exit and
proceed in the dialplan. ABE-2200 ........ ................
2010-05-17 22:10 +0000 [r263642] Mark Michelson <mmichelson@digium.com>
* /, main/devicestate.c: Merged revisions 263640 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r263640 | mmichelson | 2010-05-17 17:08:01 -0500 (Mon, 17 May
2010) | 16 lines Merged revisions 263639 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r263639 | mmichelson | 2010-05-17 17:00:28 -0500 (Mon, 17 May
2010) | 10 lines Fix logic error when checking for a devstate
provider. When using strsep, if one of the list of specified
separators is not found, it is the first parameter to strsep
which is now NULL, not the pointer returned by strsep. This issue
isn't especially severe in that the worst it is likely to do is
waste some cycles when a device with no '/' and no ':' is passed
to ast_device_state. ........ ................
2010-05-17 19:37 +0000 [r263587-263590] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, /: Merged revisions 263589 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r263589 | tilghman | 2010-05-17 14:31:15 -0500 (Mon, 17 May 2010)
| 9 lines With IMAP backend, messages in INBOX were counted twice
for MWI. (closes issue #17135) Reported by: edhorton Patches:
20100513__issue17135.diff.txt uploaded by tilghman (license 14)
17135_2.diff uploaded by ebroad (license 878) Tested by:
edhorton, ebroad ........
* main/app.c: Don't close 'n', just close 'above_n'. (closes issue
#17345) Reported by: wdoekes
2010-05-17 14:41 +0000 [r263376-263458] Leif Madsen <lmadsen@digium.com>
* main/manager.c, /: Merged revisions 263457 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r263457 | lmadsen | 2010-05-17 09:37:35 -0500 (Mon, 17 May 2010)
| 19 lines Recorded merge of revisions 263456 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r263456 | lmadsen | 2010-05-17 09:35:18 -0500 (Mon, 17 May 2010)
| 11 lines Manager cookies are not compatible with RFC2109. The
Version field in the cookies we're setting contain quotes around
the version number which is not compatible with RFC2109 and
breaks some implementations. (closes issue #17231) Reported by:
ecarruda Patches: manager_rfc2109-trunk-v1.patch uploaded by
ecarruda (license 559) manager_rfc2109-1.6.2-v1.patch uploaded by
ecarruda (license 559) Tested by: ecarruda, russell ........
................
* sounds/Makefile, /: Merged revisions 263375 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r263375 | lmadsen | 2010-05-17 09:05:33 -0500 (Mon, 17 May 2010)
| 16 lines Merged revisions 263374 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r263374 | lmadsen | 2010-05-17 09:04:57 -0500 (Mon, 17 May 2010)
| 8 lines Update link to new version of core sounds. The latest
version of the core sounds files 1.4.19 now includes the missing
queue-minute sound file which is called by app_queue but which
has been missing. (closes issue #17123) Reported by: n8ideas
........ ................
2010-05-17 13:03 +0000 [r263293] David Vossel <dvossel@digium.com>
* CHANGES, channels/chan_dahdi.c: backport of DAHDI dynamic buffer
policy dialstring option
2010-05-15 23:41 +0000 [r263202] Paul Belanger <paul.belanger@polybeacon.com>
* /, codecs/gsm/Makefile: Merged revisions 252488 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r252488 |
tilghman | 2010-03-15 12:27:08 -0400 (Mon, 15 Mar 2010) | 9 lines
Make the Makefile logic more explicit and move the Snow Leopard
logic down to where it's not executed on non-Darwin systems.
(closes issue #17028) Reported by: pabelanger Patches:
issue17028_20100315.patch uploaded by seanbright (license 71)
20100315__issue17028.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman, pabelanger ........
2010-05-13 22:13 +0000 [r263070] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 263069 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r263069 | rmudgett | 2010-05-13 17:01:36 -0500 (Thu, 13 May 2010)
| 1 line Fix inverted logic in cli command: ss7 set debug on/off
........
2010-05-13 15:36 +0000 [r262898] Russell Bryant <russell@digium.com>
* channels/chan_console.c, /: Merged revisions 262897 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r262897 | russell | 2010-05-13 10:36:12 -0500 (Thu, 13 May 2010)
| 4 lines Fix an off by one error that causes a crash. Thanks to
Raymond Burke for pointing it out. ........
2010-05-12 20:01 +0000 [r262801] Paul Belanger <paul.belanger@polybeacon.com>
* main/loader.c, main/cli.c, /: Merged revisions 262800 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r262800 | pabelanger | 2010-05-12 15:59:16 -0400 (Wed,
12 May 2010) | 8 lines Notify CLI when modules is loaded /
unloaded (closes issue #17308) Reported by: pabelanger Patches:
cli.modules.patch uploaded by pabelanger (license 224) Tested by:
pabelanger, russell ........
2010-05-12 19:53 +0000 [r262797-262799] Leif Madsen <lmadsen@digium.com>
* res/ael/pval.c, /: Merged revisions 262798 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r262798 |
lmadsen | 2010-05-12 14:53:10 -0500 (Wed, 12 May 2010) | 7 lines
Revert previous WARNING message removal. Marquis42 suggested a
better method of doing what I wanted because I ended up removing
the WARNING message for all instances when really I just wanted
to remove it for the 'return' keyword, not everything. (issue
#17145) ........
* res/ael/pval.c, /: Merged revisions 262796 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r262796 |
lmadsen | 2010-05-12 14:31:42 -0500 (Wed, 12 May 2010) | 4 lines
Remove unnecessary WARNING message in ael/pval.c (closes issue
#17145) Reported by: okrief ........
2010-05-12 18:03 +0000 [r262746] David Vossel <dvossel@digium.com>
* /, apps/app_meetme.c: Merged revisions 262744 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r262744 | dvossel | 2010-05-12 13:01:20 -0500 (Wed, 12 May 2010)
| 17 lines Merged revisions 262662 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r262662 | dvossel | 2010-05-12 12:00:04 -0500 (Wed, 12 May 2010)
| 11 lines fixes app_meetme dsp error We attempted to detect
silence after translating a frame from signed linear. This caused
a flooding of errors. To resolve this the code to detect silence
was moved before the translation. (closes issue #17133) Reported
by: jsdyer ........ ................
2010-05-12 16:29 +0000 [r262516-262659] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_privacy.c: Merged revisions 262656 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r262656 |
tilghman | 2010-05-12 11:23:26 -0500 (Wed, 12 May 2010) | 8 lines
Ensure the arguments are initialized. Also miscellaneous CG
cleanup. (closes issue #16576) Reported by: uxbod Patches:
20100505__issue16576.diff.txt uploaded by tilghman (license 14)
Tested by: uxbod ........
* /, include/asterisk/causes.h: Merged revisions 262513 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r262513 | tilghman | 2010-05-11 16:25:05 -0500 (Tue, 11
May 2010) | 7 lines Move cause 200 to cause 26, as specified in
Q.850. Also cleanup the formatting and add a few more that seem
like good candidates. (closes issue #16157) Reported by: wimpy
........
2010-05-11 19:58 +0000 [r262425] Jason Parker <jparker@digium.com>
* /, res/Makefile: Merged revisions 262422 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r262422 | qwell | 2010-05-11 14:57:24 -0500 (Tue, 11 May 2010) |
18 lines Merged revisions 262421 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r262421 | qwell | 2010-05-11 14:55:42 -0500 (Tue, 11 May 2010) |
11 lines Use a less silly method for modifying a flex-generated
file. The sed syntax that was used wasn't actually valid, causing
some versions to choke. This is the method that is used in 1.6.x+
for similar changes. (closes issue #16696) Reported by: bklang
Patches: 16696-sedfix.diff uploaded by qwell (license 4) Tested
by: qwell ........ ................
2010-05-11 19:41 +0000 [r262415-262420] Paul Belanger <paul.belanger@polybeacon.com>
* pbx/pbx_config.c, /: Merged revisions 262419 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r262419 |
pabelanger | 2010-05-11 15:40:37 -0400 (Tue, 11 May 2010) | 8
lines Improve logging by displaying line number (closes issue
#16303) Reported by: dant Patches: issue16303.patch.v2 uploaded
by pabelanger (license 224) Tested by: dant, lmadsen, pabelanger
........
* /, channels/chan_sip.c: Merged revisions 262414 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r262414 |
pabelanger | 2010-05-11 15:26:17 -0400 (Tue, 11 May 2010) | 8
lines Improve logging information for misconfigured contexts
(closes issue #17238) Reported by: pprindeville Patches:
chan_sip-bug17238.patch uploaded by pprindeville (license 347)
Tested by: pprindeville ........
2010-05-11 17:25 +0000 [r262340] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, /, Makefile.rules: Merged revisions 262330
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r262330 | tilghman | 2010-05-11 12:23:51 -0500
(Tue, 11 May 2010) | 9 lines Merged revisions 262321 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r262321 | tilghman | 2010-05-11 12:22:07 -0500 (Tue, 11
May 2010) | 2 lines Fix issue #17302 a slightly different way
(mad props to Qwell) ........ ................
2010-05-10 19:06 +0000 [r262237-262241] David Vossel <dvossel@digium.com>
* /, apps/app_directed_pickup.c: Merged revisions 262240 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r262240 | dvossel | 2010-05-10 14:06:08 -0500 (Mon, 10
May 2010) | 9 lines fixes PickupChan application (closes issue
#16863) Reported by: schern Patches: app_directed_pickup.c.patch
uploaded by schern (license 995) for_trunk.diff uploaded by
cjacobsen (license 1029) Tested by: Graber, cjacobsen, lathama,
rickead2000, dvossel ........
* channels/chan_console.c, /: Merged revisions 262236 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r262236 | dvossel | 2010-05-10 13:36:10 -0500 (Mon, 10 May 2010)
| 11 lines fixes crash in chan_console There is a race condition
between console_hangup() and start_stream(). It is possible for
console_hangup() to be called and then the stream thread to begin
after the hangup. To avoid this a check in start_stream() to make
sure the pvt-owner still exists while the pvt lock is held is
made. If the owner is gone that means the channel hung up and
start_stream should be aborted. ........
2010-05-10 16:39 +0000 [r262155] Tilghman Lesher <tlesher@digium.com>
* /, Makefile.rules: Merged revisions 262152 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r262152 | tilghman | 2010-05-10 11:36:25 -0500 (Mon, 10 May 2010)
| 17 lines Merged revisions 262151 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r262151 | tilghman | 2010-05-10 11:34:21 -0500 (Mon, 10 May 2010)
| 10 lines Allow compilation on Mac OS X 10.4 (Tiger) (closes
issue #17297) Reported by: jcovert Patches:
20100506__issue17297.diff.txt uploaded by tilghman (license 14)
(closes issue #17302) Reported by: jcovert ........
................
2010-05-09 02:17 +0000 [r261916-262105] Tilghman Lesher <tlesher@digium.com>
* autoconf/ast_ext_lib.m4, autoconf/ast_c_compile_check.m4,
autoconf/ast_c_define_check.m4, /, configure,
include/asterisk/autoconfig.h.in: Merged revisions 262102 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r262102 | tilghman | 2010-05-08 21:14:04 -0500 (Sat, 08
May 2010) | 5 lines Cleanup a bit more by getting rid of useless
version defines. Also make library detection use passed CFLAGS.
(closes issue #17309) Reported by: stuarth ........
* /, configure, configure.ac: Merged revisions 262048 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r262048 | tilghman | 2010-05-07 21:40:01 -0500 (Fri, 07 May 2010)
| 2 lines Use CPPFLAGS to pass PTHREAD_CFLAGS for vpb only
........
* /, funcs/func_odbc.c: Merged revisions 261917 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r261917 |
tilghman | 2010-05-07 15:54:35 -0500 (Fri, 07 May 2010) | 8 lines
Double free crash (closes issue #17245) Reported by:
thedavidfactor Patches: 20100426__issue17245.diff.txt uploaded by
tilghman (license 14) Tested by: murraytm ........
* /, configure, include/asterisk/autoconfig.h.in, configure.ac:
Merged revisions 261913 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r261913 |
tilghman | 2010-05-07 15:35:17 -0500 (Fri, 07 May 2010) | 14
lines Use the detected pthread building flags in every place,
instead of hardcoding -lpthread. We nicely detect the right flags
on each system for building Asterisk with pthreads, then ignore
it for every other build option that requires us to build with
pthreads. This caused some items to return a false negative. Also
cleanup some minor naming issues that caused "library library"
redundancy in the output. (closes issue #17303) Reported by:
stuarth Patches: 20100507__issue17303.diff.txt uploaded by
tilghman (license 14) Tested by: stuarth ........
2010-05-07 16:08 +0000 [r261868] Leif Madsen <lmadsen@digium.com>
* UPGRADE-1.6.txt, /: Merged revisions 261867 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r261867 |
lmadsen | 2010-05-07 11:05:24 -0500 (Fri, 07 May 2010) | 6 lines
Update UPGRADE-1.6.txt stating insecure=very has been removed.
(closes issue #17282) Reported by: stuarth Tested by: stuarth
........
2010-05-06 20:13 +0000 [r261739] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 261736 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r261736 | jpeeler | 2010-05-06 15:11:53 -0500
(Thu, 06 May 2010) | 15 lines Merged revisions 261735 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r261735 | jpeeler | 2010-05-06 15:10:59 -0500 (Thu, 06 May 2010)
| 8 lines Only allow the operator key to be accepted after
leaving a voicemail. Or rather disallow the operator key from
being accepted when not offered, such as after finishing a
recording from within the mailbox options menu. ABE-2121 SWP-1267
........ ................
2010-05-06 17:08 +0000 [r261612] Jason Parker <jparker@digium.com>
* sounds/Makefile, /: Merged revisions 261609 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r261609 | qwell | 2010-05-06 12:06:40 -0500 (Thu, 06 May 2010) |
11 lines Merged revisions 261608 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r261608 | qwell | 2010-05-06 11:56:02 -0500 (Thu, 06 May 2010) |
4 lines Use the versioned MOH tarballs, now that we have them.
This makes for more reproducibility. Prompted by a discussion in
#asterisk-dev ........ ................
2010-05-06 15:43 +0000 [r261563] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c: Merged revisions 261560 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r261560 |
tilghman | 2010-05-06 10:39:10 -0500 (Thu, 06 May 2010) | 8 lines
Permit more lines within a SIP body to be parsed. The example
given within the related issue showed 120 lines, which was mostly
a result of the body being XML. (closes issue #17179) Reported
by: khw ........
2010-06-01 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.8 Released.
2010-05-26 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.8-rc2 Released.
2010-05-26 10:56 -0500 [r265891] Matt Nicholson <mnicholson@digium.com>
* Merged r265610 from 1.4:
Don't mark the cdr records of unanswered queue calls with "NOANSWER".
This restores the behavior prior to r258670.
(closes issue #17334)
Reported by: jvandal
Patches:
queue-cdr-fixes1.diff uploaded by mnicholson (license 96)
Tested by: aragon, jvandal
2010-05-06 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.8-rc1 Released
2010-05-06 14:07 +0000 [r261498-261499] Russell Bryant <russell@digium.com>
* tests/test_heap.c: Add test case that ensures the heap handles
arbitrary removals properly. (issue #17277) Reported by:
cappucinoking Patches: test_heap.diff uploaded by cappucinoking
(license 1036) Tested by: cappucinoking, russell
* /, main/heap.c: Merged revisions 261496 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r261496 |
russell | 2010-05-06 08:58:07 -0500 (Thu, 06 May 2010) | 40 lines
Fix handling of removing nodes from the middle of a heap. This
bug surfaced in 1.6.2 and does not affect code in any other
released version of Asterisk. It manifested itself as SIP qualify
not happening when it should, causing peers to go unreachable.
This was debugged down to scheduler entries sometimes not getting
executed when they were supposed to, which was in turn caused by
an error in the heap code. The problem only sometimes occurs, and
it is due to the logic for removing an entry in the heap from an
arbitrary location (not just popping off the top). The scheduler
performs this operation frequently when entries are removed
before they run (when ast_sched_del() is used). In a normal pop
off of the top of the heap, a node is taken off the bottom,
placed at the top, and then bubbled down until the max heap
property is restored (see max_heapify()). This same logic was
used for removing an arbitrary node from the middle of the heap.
Unfortunately, that logic is full of fail. This patch fixes that
by fully restoring the max heap property when a node is thrown
into the middle of the heap. Instead of just pushing it down as
appropriate, it first pushes it up as high as it will go, and
_then_ pushes it down. Lastly, fix a minor problem in
ast_heap_verify(), which is only used for debugging. If a parent
and child node have the same value, that is not an error. The
only error is if a parent's value is less than its children. A
huge thanks goes out to cappucinoking for debugging this down to
the scheduler, and then producing an ast_heap test case that
demonstrated the breakage. That made it very easy for me to focus
on the heap logic and produce a fix. Open source projects are
awesome. (closes issue #16936) Reported by: ib2 Tested by:
cappucinoking, crjw (closes issue #17277) Reported by:
cappucinoking Patches: heap-fix.rev2.diff uploaded by russell
(license 2) Tested by: cappucinoking, russell ........
2010-05-06 07:43 +0000 [r261453] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* channels/chan_dahdi.c, /: Merged revisions 261451 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r261451 | tzafrir | 2010-05-06 10:27:31 +0300 (ה', 06 מאי 2010) |
4 lines When failing to configure, don't destroy 'cfg' twice
Fixes a crash when some config section had an incorrect channel
config. ........
2010-05-05 19:08 +0000 [r261233-261315] Paul Belanger <paul.belanger@polybeacon.com>
* /, channels/chan_sip.c: Merged revisions 261314 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r261314 | pabelanger | 2010-05-05 14:43:03 -0400 (Wed, 05 May
2010) | 19 lines Merged revisions 261274 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r261274 | pabelanger | 2010-05-05 12:42:22 -0400 (Wed, 05 May
2010) | 12 lines Registration fix for SIP realtime. Make sure
realtime fields are not empty. (closes issue #17266) Reported by:
Nick_Lewis Patches: chan_sip.c-realtime.patch uploaded by Nick
Lewis (license 657) Tested by: Nick_Lewis, sberney Review:
https://reviewboard.asterisk.org/r/643/ ........ ................
* apps/app_queue.c, /: Merged revisions 261232 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r261232 |
pabelanger | 2010-05-05 11:42:07 -0400 (Wed, 05 May 2010) | 10
lines 'queue reset stats' erroneously clears wrapuptime
configuration. Resets each member's lastcall to 0 now. (closes
issue #17262, #16519) Reported by: rain Patches:
wrapuptime_reset_fix.diff uploaded by rain (license 327) Tested
by: rain ........
2010-05-04 23:55 +0000 [r261098] Tilghman Lesher <tlesher@digium.com>
* main/channel.c, /: Merged revisions 261095 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r261095 | tilghman | 2010-05-04 18:51:52 -0500 (Tue, 04 May 2010)
| 18 lines Merged revisions 261093-261094 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r261093 | tilghman | 2010-05-04 18:36:53 -0500 (Tue, 04 May 2010)
| 7 lines Protect against overflow, when calculating how long to
wait for a frame. (closes issue #17128) Reported by: under
Patches: d.diff uploaded by under (license 914) ........ r261094
| tilghman | 2010-05-04 18:47:08 -0500 (Tue, 04 May 2010) | 2
lines Add a tiny corner case to the previous commit ........
................
2010-05-04 19:01 +0000 [r260927] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 260924 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r260924 | jpeeler | 2010-05-04 13:51:28 -0500
(Tue, 04 May 2010) | 18 lines Merged revisions 260923 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r260923 | jpeeler | 2010-05-04 13:46:46 -0500 (Tue, 04 May 2010)
| 12 lines Voicemail transfer to operator should occur
immediately, not after main menu. There were two scenarios in the
advanced options that while using the operator=yes and review=yes
options, the transfer occurred only after exiting the main menu
(after sending a reply or leaving a message for an extension).
Now after the audio is processed for the reply or message the
transfer occurs immediately as expected. ABE-2107 ABE-2108
........ ................
2010-05-04 16:58 +0000 [r260884] Matthew Nicholson <mnicholson@digium.com>
* configs/sip.conf.sample, include/asterisk/frame.h,
main/channel.c, /, channels/chan_sip.c: Merged revisions 254450
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r254450 | kpfleming | 2010-03-25 10:27:31 -0500 (Thu, 25
Mar 2010) | 49 lines Improve handling of T.38 re-INVITEs that
arrive before a T.38-capable application is executing on a
channel. This patch addresses an issue found during working with
end-users using res_fax. If an incoming call is answered in the
dialplan, or jumps to the 'fax' extension due to reception of a
CNG tone (with faxdetect enabled), and then the remote endpoint
sends a T.38 re-INVITE, it is possible for the channel's T.38
state to be 'T38_STATE_NEGOTIATING' when the application starts
up. Unfortunately, even if the application wants to use T.38, it
can't respond to the peer's negotiation request, because the
AST_CONTROL_T38_PARAMETERS control frame that chan_sip sent
originally has been lost, and the application needs the content
of that frame to be able to formulate a reply. This patch adds a
new 'request' type to AST_CONTROL_T38_PARAMETERS,
AST_T38_REQUEST_PARMS. If the application sends this request,
chan_sip will re-send the original control frame (with
AST_T38_REQUEST_NEGOTIATE as the request type), and the
application can respond as normal. If this occurs within the five
second timeout in chan_sip, the automatic cancellation of the
peer reinvite will be stopped, and the application will 'own' the
negotiation process from that point onwards. This also improves
the code path in chan_sip to allow sip_indicate(), when called
for AST_CONTROL_T38_PARAMETERS, to be able to return a non-zero
response, which should have been in place before since the
control frame *can* fail to be processed properly. It also
modifies ast_indicate() to return whatever result the channel
driver returned for this control frame, rather than converting
all non-zero results into '-1'. Finally, the new request type
intentionally returns a positive value, so that an application
that sends AST_T38_REQUEST_PARMS can know for certain whether the
channel driver accepted it and will be replying with a control
frame of its own, or whether it was ignored (if the
sip_indicate()/ast_indicate() path had properly supported failure
responses before, this would not be necessary). This patch also
modifies res_fax to take advantage of the new request. In
addition, this patch makes sip_t38_abort() actually lock the
private structure before doing its work... bad programmer, no
donut. This patch also enhances chan_sip's 'faxdetect' support to
allow triggering on T.38 re-INVITEs received as well as CNG tone
detection. Review: https://reviewboard.asterisk.org/r/556/
........
2010-05-04 15:51 +0000 [r260746-260805] Jason Parker <jparker@digium.com>
* /, build_tools/make_build_h: Merged revisions 260802 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r260802 | qwell | 2010-05-04 10:49:57 -0500
(Tue, 04 May 2010) | 9 lines Merged revisions 260801 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r260801 | qwell | 2010-05-04 10:49:27 -0500 (Tue, 04 May
2010) | 1 line Fix fallout from removing from configure script.
Pointed out by philipp64 on #asterisk-dev ........
................
* /: Fix merge props
2010-05-03 17:42 +0000 [r260743] Paul Belanger <paul.belanger@polybeacon.com>
* Makefile, /: Merged revisions 260661-260662 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r260661 | pabelanger | 2010-05-03 12:41:30 -0400 (Mon, 03 May
2010) | 10 lines non-root make install PREFIX=/tmp fails. Prepend
libdir when executing mkpkgconfig allowing non-root installs to
work. (closes issue #17268) Reported by: pabelanger Patches:
issue17268.patch uploaded by pabelanger (license 224) Tested by:
pabelanger ........ r260662 | pabelanger | 2010-05-03 12:54:41
-0400 (Mon, 03 May 2010) | 3 lines Should have removed /usr/lib/
part. Thanks Qwell. ........
2010-05-03 14:59 +0000 [r260571] Leif Madsen <lmadsen@digium.com>
* doc/HOWTO_collect_debug_information.txt: Merged revisions 260570
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r260570 | lmadsen | 2010-05-03 09:58:23 -0500
(Mon, 03 May 2010) | 9 lines Merged revisions 260569 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r260569 | lmadsen | 2010-05-03 09:57:39 -0500 (Mon, 03
May 2010) | 1 line Minor typo pointed out by pabelanger on IRC.
........ ................
2010-04-30 22:48 +0000 [r260441] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 260437 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r260437 | jpeeler | 2010-04-30 17:36:49 -0500
(Fri, 30 Apr 2010) | 18 lines Merged revisions 260434 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r260434 | jpeeler | 2010-04-30 17:22:46 -0500 (Fri, 30 Apr 2010)
| 11 lines Ensure channel state is not incorrectly set in the
case of a very early answer. The needringing bit was being read
in dahdi_read after answering thereby setting the state to
ringing from up. This clears needringing upon answering so that
is no longer possible. (closes issue #17067) Reported by: tzafrir
Patches: needringing.diff uploaded by tzafrir (license 46)
........ ................
2010-04-30 20:22 +0000 [r260373] Mark Michelson <mmichelson@digium.com>
* res/res_musiconhold.c, /: Merged revisions 260346 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r260346 | mmichelson | 2010-04-30 15:11:02 -0500
(Fri, 30 Apr 2010) | 24 lines Merged revisions 260345 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r260345 | mmichelson | 2010-04-30 15:08:15 -0500 (Fri, 30 Apr
2010) | 18 lines Fix potential crash from race condition due to
accessing channel data without the channel locked. In
res_musiconhold.c, there are several places where a channel's
stream's existence is checked prior to calling ast_closestream on
it. The issue here is that in several cases, the channel was not
locked while checking the stream. The result was that if two
threads checked the state of the channel's stream at
approximately the same time, then there could be a situation
where both threads attempt to call ast_closestream on the
channel's stream. The result here is that the refcount for the
stream would go below 0, resulting in a crash. I have added
proper channel locking to res_musiconhold.c to ensure that we do
not try to check chan->stream without the channel locked. A
Digium customer has been using this patch for several weeks and
has not had any crashes since applying the patch. ABE-2147
........ ................
2010-04-30 06:22 +0000 [r260281-260303] Tilghman Lesher <tlesher@digium.com>
* /, main/app.c: Merged revisions 260292 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r260292 |
tilghman | 2010-04-30 01:19:35 -0500 (Fri, 30 Apr 2010) | 13
lines Don't allow file descriptors to go above 64k, when we're
closing them in a fork(2). This saves time, when, even though the
system allows the process limit to be that high, the practical
limit is much lower. (closes issue #17223) Reported by:
dbackeberg Patches: 20100423__issue17223.diff.txt uploaded by
tilghman (license 14) Tested by: dbackeberg ........
* configs/extensions.conf.sample, /: Merged revisions 260280 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r260280 | tilghman | 2010-04-30 00:23:56 -0500 (Fri, 30
Apr 2010) | 7 lines Logic fixups for a sample FREENUM dialplan
context. (closes issue #17263) Reported by: pprindeville Patches:
freenum-dialplan.patch#3 uploaded by pprindeville (license 347)
........
2010-04-29 23:13 +0000 [r260234] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 260231 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r260231 | rmudgett | 2010-04-29 17:44:14 -0500
(Thu, 29 Apr 2010) | 33 lines Merged revisions 260195 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r260195 | rmudgett | 2010-04-29 17:11:47 -0500 (Thu, 29 Apr 2010)
| 26 lines DTMF CallerID detection problems. The code handling
DTMF CallerID drops digits on long CallerID numbers and may
timeout waiting for the first ring with shorter numbers. The DTMF
emulation mode was not turned off when processing DTMF CallerID.
When the emulation code gets behind in processing the DTMF digits
it can skip a digit. For shorter numbers, the timeout may have
been too short. I increased it from 2 seconds to 4 seconds. Four
seconds is a typical time between rings for many countries.
(closes issue #16460) Reported by: sum Patches: issue16460.patch
uploaded by rmudgett (license 664) issue16460_v1.6.2.patch
uploaded by rmudgett (license 664) Tested by: sum, rmudgett
Review: https://reviewboard.asterisk.org/r/634/ JIRA SWP-562 JIRA
AST-334 JIRA SWP-901 ........ ................
2010-04-29 18:18 +0000 [r260156] Tilghman Lesher <tlesher@digium.com>
* configs/extensions.conf.sample, /: Merged revisions 260148 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r260148 | tilghman | 2010-04-29 13:15:57 -0500 (Thu, 29
Apr 2010) | 2 lines Pattern match fail. ........
2010-04-29 15:35 +0000 [r260051] David Vossel <dvossel@digium.com>
* main/audiohook.c, /, include/asterisk/audiohook.h: Merged
revisions 260050 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r260050 | dvossel | 2010-04-29 10:33:27 -0500 (Thu, 29 Apr 2010)
| 21 lines Merged revisions 260049 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r260049 | dvossel | 2010-04-29 10:31:02 -0500 (Thu, 29 Apr 2010)
| 14 lines Fixes crash in audiohook_write_list The middle_frame
in the audiohook_write_list function was being freed if a
audiohook manipulator returned a failure. This is incorrect
logic. This patch resolves this and adds detailed descriptions of
how this function should work and why manipulator failures must
be ignored. (closes issue #17052) Reported by: dvossel Tested by:
dvossel (closes issue #16196) Reported by: atis Review:
https://reviewboard.asterisk.org/r/623/ ........ ................
2010-04-28 22:36 +0000 [r259959] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 259957 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r259957 |
mmichelson | 2010-04-28 17:34:15 -0500 (Wed, 28 Apr 2010) | 11
lines Don't override peer context with domain context. (closes
issue #17040) Reported by: pprindeville Patches:
asterisk-1.6-bugid17040.patch uploaded by pprindeville (license
347) Tested by: pprindeville Review:
https://reviewboard.asterisk.org/r/565/ ........
2010-04-28 21:26 +0000 [r259899] David Vossel <dvossel@digium.com>
* main/channel.c, channels/chan_local.c, /: Merged revisions 259870
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r259870 | dvossel | 2010-04-28 16:20:03 -0500
(Wed, 28 Apr 2010) | 39 lines Merged revisions 259858 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259858 | dvossel | 2010-04-28 16:16:03 -0500 (Wed, 28 Apr 2010)
| 33 lines resolves deadlocks in chan_local Issue_1. In the
local_hangup() 3 locks must be held at the same time... pvt,
pvt->chan, and pvt->owner. Proper deadlock avoidance is done when
the channel to hangup is the outbound chan_local channel, but
when it is not the outbound channel we have an issue... We
attempt to do deadlock avoidance only on the tech pvt, when both
the tech pvt and the pvt->owner are locked coming into that loop.
By never giving up the pvt->owner channel deadlock avoidance is
not entirely possible. This patch resolves that by doing deadlock
avoidance on both the pvt->owner and the pvt when trying to get
the pvt->chan lock. Issue_2. ast_prod() is used in
ast_activate_generator() to queue a frame on the channel and make
the channel's read function get called. This function is used in
ast_activate_generator() while the channel is locked, which
mean's the channel will have a lock both from the generator code
and the frame_queue code by the time it gets to chan_local.c's
local_queue_frame code... local_queue_frame contains some of the
same crazy deadlock avoidance that local_hangup requires, and
this recursive lock prevents that deadlock avoidance from
happening correctly. This patch removes ast_prod() from the
channel lock so only one lock is held during the
local_queue_frame function. (closes issue #17185) Reported by:
schmoozecom Patches: issue_17185_v1.diff uploaded by dvossel
(license 671) issue_17185_v2.diff uploaded by dvossel (license
671) Tested by: schmoozecom, GameGamer43 Review:
https://reviewboard.asterisk.org/r/631/ ........ ................
2010-04-28 21:09 +0000 [r259854] Leif Madsen <lmadsen@digium.com>
* config.guess: Merged revisions 259853 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r259853 | lmadsen | 2010-04-28 16:08:34 -0500 (Wed, 28 Apr 2010)
| 14 lines Merged revisions 259852 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259852 | lmadsen | 2010-04-28 16:07:48 -0500 (Wed, 28 Apr 2010)
| 6 lines Update config.guess. Updating config.guess because
after installing Ubuntu Server 9.10 and running all the update
scripts, running ./configure would not continue because it was
unable to determine what kind of system I had. After updating
config.guess things started working again. ........
................
2010-04-28 20:34 +0000 [r259781-259851] Jason Parker <jparker@digium.com>
* /, configure, configure.ac: Merged revisions 259848 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r259848 | qwell | 2010-04-28 15:32:14 -0500
(Wed, 28 Apr 2010) | 9 lines Merged revisions 259847 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r259847 | qwell | 2010-04-28 15:30:21 -0500 (Wed, 28 Apr
2010) | 1 line Add AC_CONFIG_AUX_DIR to configure script, so
systems without install can use install-sh from our source dir.
........ ................
* makeopts.in, /: Merged revisions 259837 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r259837 | qwell | 2010-04-28 15:26:35 -0500 (Wed, 28 Apr 2010) |
9 lines Merged revisions 259833 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259833 | qwell | 2010-04-28 15:25:36 -0500 (Wed, 28 Apr 2010) |
1 line Missed this when removing $ID ........ ................
* Makefile, /, configure, configure.ac: Merged revisions 259760 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r259760 | qwell | 2010-04-28 14:19:54 -0500
(Wed, 28 Apr 2010) | 14 lines Merged revisions 259748 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259748 | qwell | 2010-04-28 14:17:38 -0500 (Wed, 28 Apr 2010) |
7 lines Remove usage of `id` since it isn't useful and was
causing breakge. Solaris `id` doesn't support the -u argument.
Instead of figuring out how to fix this to work on Solaris, I
decided to check why it was necessary and where else it was used.
It was only used in one place, and it hasn't been needed for a
very long time (I question whether it was ever needed). ........
................
2010-04-28 17:19 +0000 [r259681] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 259672 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r259672 | jpeeler | 2010-04-28 12:18:43 -0500
(Wed, 