2010-05-26 Leif Madsen <lmadsen@digium.com>
* 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, 28 Apr 2010) | 11 lines Merged revisions 259664 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259664 | jpeeler | 2010-04-28 12:13:29 -0500 (Wed, 28 Apr 2010)
| 4 lines Do not play goodbye prompt after timeout of message
review. ABE-2124 ........ ................
2010-04-27 22:46 +0000 [r259616] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 259538 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r259538 | rmudgett | 2010-04-27 17:18:09 -0500
(Tue, 27 Apr 2010) | 18 lines Merged revisions 259531 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259531 | rmudgett | 2010-04-27 16:53:07 -0500 (Tue, 27 Apr 2010)
| 11 lines DAHDI "WARNING" message is confusing and vague
"WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed
failed: Success" Changed the warning to "Failed to decode
CallerID on channel 'name'". The message before it is likely more
specific about why the CallerID decode failed. SWP-501 AST-283
........ ................
2010-04-27 21:50 +0000 [r259528] Leif Madsen <lmadsen@digium.com>
* sounds/Makefile: Merged revisions 259527 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r259527 | lmadsen | 2010-04-27 16:49:36 -0500 (Tue, 27 Apr 2010)
| 23 lines Merged revisions 259526 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259526 | lmadsen | 2010-04-27 16:48:47 -0500 (Tue, 27 Apr 2010)
| 15 lines Update sounds files. * Add additional sounds prompts
for say_enumeration * Update the English conference sounds
prompts so they are better quality and all sound more consistent
* Clean up the core-sounds-XX.txt and extra-sounds-XX.txt files
to include all present sound files Both core (en, fr, es) and
extra (en, fr) sounds files have been updated. (closes issue
#16200) Reported by: murf (closes issue #17137) Reported by:
lmadsen ........ ................
2010-04-27 21:25 +0000 [r259356-259486] Jason Parker <jparker@digium.com>
* main/editline/configure.in, /, main/editline/configure,
main/editline/Makefile.in: Merged revisions 259439 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r259439 | qwell | 2010-04-27 16:13:01 -0500 (Tue, 27 Apr 2010) |
5 lines Add gar to the check for AR for those silly OSes
(Solaris) that don't have ar. autoconf2.13 couldn't handle
AC_PROG_GREP, so I removed it. This is fine, since we don't need
to use anything that the configure script doesn't. ........
* /: Unblock revision 259439.
* /, configure, configure.ac: Merged revisions 259353 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r259353 | qwell | 2010-04-27 14:31:55 -0500
(Tue, 27 Apr 2010) | 12 lines Merged revisions 259352 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259352 | qwell | 2010-04-27 14:29:26 -0500 (Tue, 27 Apr 2010) |
5 lines Support the silly OSes that don't have ar and strip.
Since AC_PATH_TOOL is equiv to AC_CHECK_TOOL when path isn't
specified, and AC_PATH_TOOLS doesn't exist, we'll just switch to
AC_CHECK_TOOLS. ........ ................
2010-04-27 19:03 +0000 [r259310] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged
revisions 259307 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r259307 | rmudgett | 2010-04-27 13:29:33 -0500 (Tue, 27 Apr 2010)
| 21 lines Merged revisions 259270 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010)
| 14 lines hidecalleridname parameter in chan_dahdi.conf Issue
#7321 implements a new chan_dahdi configuration option. However,
a change mentioned in the issue was never implemented. This is
the change that will allow the feature to work. I added a note to
chan_dahdi.conf.sample about the feature. (closes issue #17143)
Reported by: djensen99 Patches: diff.txt uploaded by djensen99
(license NA) (One line change) Tested by: djensen99 ........
................
2010-04-26 21:48 +0000 [r259103-259109] Mark Michelson <mmichelson@digium.com>
* main/channel.c, /: Merged revisions 259105 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r259105 | mmichelson | 2010-04-26 16:45:13 -0500 (Mon, 26 Apr
2010) | 9 lines Merged revisions 259104 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259104 | mmichelson | 2010-04-26 16:44:43 -0500 (Mon, 26 Apr
2010) | 3 lines Let compilation succeed warning-free when
DONT_OPTIMIZE is turned off. ........ ................
* main/channel.c, /: Merged revisions 259023 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r259023 | mmichelson | 2010-04-26 16:13:35 -0500 (Mon, 26 Apr
2010) | 19 lines Merged revisions 259018 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259018 | mmichelson | 2010-04-26 16:03:08 -0500 (Mon, 26 Apr
2010) | 13 lines Prevent Newchannel manager events for dummy
channels. No Newchannel manager event will be fired for channels
that are allocated to not match a registered technology type.
Thus bogus channels allocated solely for variable substitution or
CDR operations do not result in a Newchannel event. (closes issue
#16957) Reported by: atis Review:
https://reviewboard.asterisk.org/r/601 ........ ................
2010-04-26 16:00 +0000 [r258935] Leif Madsen <lmadsen@digium.com>
* /, channels/chan_sip.c: Merged revisions 258934 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r258934 |
lmadsen | 2010-04-26 10:59:34 -0500 (Mon, 26 Apr 2010) | 7 lines
Small error in the T.140 RTP port verbose log. (closes issue
#16988) Reported by: frawd Patches: chan_sip_sdp_verbose_fix.diff
uploaded by frawd (license 610) Tested by: russell ........
2010-04-25 18:14 +0000 [r258779] Tilghman Lesher <tlesher@digium.com>
* res/res_monitor.c, /: Merged revisions 258776 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r258776 | tilghman | 2010-04-25 13:12:14 -0500 (Sun, 25 Apr 2010)
| 13 lines Merged revisions 258775 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r258775 | tilghman | 2010-04-25 13:09:05 -0500 (Sun, 25 Apr 2010)
| 6 lines When StopMonitor is called, ensure that it will not be
restarted by a channel event. (closes issue #16590) Reported by:
kkm Patches: resmonitor-16590-trunk.239289.diff uploaded by kkm
(license 888) ........ ................
2010-04-22 22:15 +0000 [r258676] Matthew Nicholson <mnicholson@digium.com>
* main/cdr.c, main/channel.c, /, main/features.c: Merged revisions
258671,258675 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r258671 | mnicholson | 2010-04-22 16:57:59 -0500 (Thu, 22 Apr
2010) | 32 lines Merged revisions 193391,258670 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r193391 | mnicholson | 2009-05-08 16:01:25 -0500 (Fri, 08 May
2009) | 8 lines Set the proper disposition on originated calls.
(closes issue #14167) Reported by: jpt Patches:
call-file-missing-cdr2.diff uploaded by mnicholson (license 96)
Tested by: dlotina, rmartinez, mnicholson ........ r258670 |
mnicholson | 2010-04-22 16:49:07 -0500 (Thu, 22 Apr 2010) | 11
lines Fix broken CDR behavior. This change allows a CDR record
previously marked with disposition ANSWERED to be set as BUSY or
NO ANSWER. Additionally this change partially reverts r235635 and
does not set the AST_CDR_FLAG_ORIGINATED flag on CDRs generated
from ast_call(). To preserve proper CDR behavior, the
AST_CDR_FLAG_DIALED flag is now cleared from all brige CDRs in
ast_bridge_call(). (closes issue #16797) Reported by:
VarnishedOtter Tested by: mnicholson ........ (closes issue
#16222) Reported by: telles Tested by: mnicholson
................ r258675 | mnicholson | 2010-04-22 17:11:23 -0500
(Thu, 22 Apr 2010) | 2 lines Fix previous commit.
................
2010-04-22 21:58 +0000 [r258516-258672] Russell Bryant <russell@digium.com>
* /, main/event.c: Merged revisions 258632 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk For 1.6.2, only
merge the bug fixes, not the unit test. ........ r258632 |
russell | 2010-04-22 16:06:53 -0500 (Thu, 22 Apr 2010) | 22 lines
Add ast_event subscription unit test and fix some ast_event API
bugs. This patch introduces another test in test_event.c that
exercises most of the subscription related ast_event API calls. I
made some minor additions to the existing event allocation test
to increase API coverage by the test code. Finally, I made a list
in a comment of API calls not yet touched by the test module as a
to-do list for future test development. During the development of
this test code, I discovered a number of bugs in the event API.
1) subscriptions to AST_EVENT_ALL were not handled appropriately
in a couple of different places. The API allows a subscription to
all event types, but with IE parameters, just as if it was a
subscription to a specific event type. However, the parameters
were being ignored. This affected ast_event_check_subscriber()
and event distribution to subscribers. 2) Some of the logic in
ast_event_check_subscriber() for checking subscriptions against
query parameters was wrong. Review:
https://reviewboard.asterisk.org/r/617/ ........
* /, doc/tex/channelvariables.tex: Merged revisions 258515 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r258515 | russell | 2010-04-22 12:36:34 -0500 (Thu, 22
Apr 2010) | 2 lines Add MEETMEBOOKID from r256019. ........
2010-04-21 22:11 +0000 [r258436] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 258433 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r258433 | jpeeler | 2010-04-21 16:56:09 -0500
(Wed, 21 Apr 2010) | 15 lines Merged revisions 258432 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r258432 | jpeeler | 2010-04-21 16:45:36 -0500 (Wed, 21 Apr 2010)
| 8 lines Fix looping forever when no input received in certain
voicemail menu scenarios. Specifically, prompting for an
extension (when leaving or forwarding a message) or when
prompting for a digit (when saving a message or changing
folders). ABE-2122 SWP-1268 ........ ................
2010-04-21 19:44 +0000 [r258384-258386] Leif Madsen <lmadsen@digium.com>
* doc/tex/asterisk.tex: Remove missed line in previous merge.
(issue #17220)
* configure: Forgot to merge the updated configure script. (issue
#17220)
* doc/tex/localchannel.tex, doc/tex/enum.tex, makeopts.in,
doc/tex/asterisk.tex, Makefile, /, doc/tex/Makefile,
configure.ac, doc/tex/phoneprov.tex, doc/tex, doc/tex/ael.tex,
build_tools/prep_tarball: Merged revisions 258351 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r258351 | lmadsen | 2010-04-21 14:18:35 -0500 (Wed, 21 Apr 2010)
| 20 lines Add ability to generate ASCII documentation from the
TeX files. These changes add the ability to run 'make
asterisk.txt' just like the existing 'make asterisk.pdf' commands
to generate a text document from the TeX files we have in the
doc/tex/ directory. I've also updated a few of the .tex files
because they weren't properly escaping certain characters so they
would show up as Unicode characters (like [U+021C]). Made changes
to the configure scripts so it would detect the catdvi program
which is required to convert the .dvi file generated by latex.
I've also added a few lines to the build_tools/prep_tarball
script so that the text documentation gets generated and added to
future tarballs of Asterisk releases. (closes issue #17220)
Reported by: lmadsen Patches: asterisk.txt.patch uploaded by
lmadsen (license 10) asterisk.txt.patch-v4 uploaded by pabelanger
(license 224) Tested by: lmadsen, pabelanger ........
2010-04-21 18:19 +0000 [r258314] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 258305 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r258305 |
dvossel | 2010-04-21 13:13:36 -0500 (Wed, 21 Apr 2010) | 12 lines
fixes issue with double "sip:" in header field This is a clear
mistake in logic. Future discussions about how to avoid having to
handle uri's like this should take place in the future, but this
fix needs to go in for now. (closes issue #15847) Reported by:
ebroad Patches: doublesip.patch uploaded by ebroad (license 878)
........
2010-04-20 19:03 +0000 [r258148-258150] Leif Madsen <lmadsen@digium.com>
* /, configs/cli_aliases.conf.sample: Merged revisions 258149 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r258149 | lmadsen | 2010-04-20 14:02:49 -0500 (Tue, 20
Apr 2010) | 1 line Add 'soft hangup' alias per Steve Johnson on
asterisk-users. ........
* configs/extensions.conf.sample, /: Merged revisions 258147 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r258147 | lmadsen | 2010-04-20 13:38:39 -0500 (Tue, 20
Apr 2010) | 8 lines Add example dialplan for dialing ISN numbers
(http://www.freenum.org). Minor tweaks and documentation added by
me. (closes issue #17058) Reported by: pprindeville Patches:
freenum.patch#5 uploaded by pprindeville (license 347) Tested by:
lmadsen ........
2010-04-20 18:04 +0000 [r258108] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 258065 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r258065 | jpeeler | 2010-04-20 12:06:19 -0500
(Tue, 20 Apr 2010) | 17 lines Merged revisions 258029 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r258029 | jpeeler | 2010-04-20 11:16:33 -0500 (Tue, 20 Apr 2010)
| 11 lines Play correct prompt when voicemail store failure
occurs after attempted forward. If a user's mailbox was full and
a message was attempted to be forwarded to said box, warnings on
the console would indicate failure. However, the played prompt
was that of success (vm-msgsaved). Now storage failure is taken
into account and the correct prompt (vm-mailboxfull) is played
when appropriate. ABE-2123 SWP-1262 ........ ................
2010-04-20 18:02 +0000 [r258107] Leif Madsen <lmadsen@digium.com>
* contrib/scripts/sip-friends.sql, /: Merged revisions 258106 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r258106 | lmadsen | 2010-04-20 13:01:28 -0500 (Tue, 20
Apr 2010) | 7 lines Add missing 'useragent' field to
sip-friends.sql file. (closes issue #17171) Reported by: thehar
Patches: sip-friends.patch uploaded by thehar (license 831)
Tested by: pabelanger, thehar ........
2010-04-19 21:58 +0000 [r257948-257950] Jason Parker <jparker@digium.com>
* main/indications.c, /: Merged revisions 257949 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r257949 |
qwell | 2010-04-19 16:57:56 -0500 (Mon, 19 Apr 2010) | 1 line
Change log message to match severity. ........
* main/indications.c, /: Merged revisions 257947 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r257947 |
qwell | 2010-04-19 16:49:30 -0500 (Mon, 19 Apr 2010) | 6 lines
Don't consider a missing indications.conf to be a critical error.
There were many changes in revision 176627 which would avoid the
error that a missing config would have caused. Other than this,
there are no other config files (including asterisk.conf,
surprisingly) that are required. ........
2010-04-19 18:30 +0000 [r257850] Terry Wilson <twilson@digium.com>
* /, main/features.c: Merged revisions 257810 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r257810 |
twilson | 2010-04-19 12:57:41 -0500 (Mon, 19 Apr 2010) | 5 lines
Fix incomplete CDR merge from r195881 Because res/res_features.c
was removed and main/cdr.c added, these changes didn't make it to
trunk and the 1.6.x branches ........
2010-04-18 17:28 +0000 [r257771] Tilghman Lesher <tlesher@digium.com>
* configs/cdr_odbc.conf.sample, /: Merged revisions 257768 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r257768 | tilghman | 2010-04-18 12:25:53 -0500 (Sun, 18
Apr 2010) | 2 lines Removing unused configuration parameters
........
2010-04-16 21:47 +0000 [r257740] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
* apps/app_mixmonitor.c, /: Merged revisions 257713 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r257713 | dhubbard | 2010-04-16 16:22:30 -0500
(Fri, 16 Apr 2010) | 28 lines Merged revisions 257686 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r257686 | dhubbard | 2010-04-16 16:15:43 -0500 (Fri, 16 Apr 2010)
| 21 lines Make the mixmonitor thread process audio frames faster
Mantis issue 17078 reports MixMonitor recordings have shorter
durations than the call duration. This was because the mixmonitor
thread was not processing frames from the audiohook fast enough.
The mixmonitor thread would slowly fall behind the most recent
audio frame and when the channel hangs up, the mixmonitor thread
would exit without processing the same number of frames as the
channel; leaving the mixmonitor recording shorter than actual
call duration. This revision fixes this issue by moving the
ast_audiohook_trigger_wait() and the subsequent audiohook.status
check into the block where the ast_audiohook_read_frame()
function returns NULL. (closes issue #17078) Reported by:
geoff2010 Patches: dw-M17078.patch uploaded by dhubbard (license
733) Tested by: dhubbard, geoff2010 Review:
https://reviewboard.asterisk.org/r/611/ ........ ................
2010-04-15 21:34 +0000 [r257510-257597] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/app.h, /, main/app.c: Merged revisions 257560
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r257560 | tilghman | 2010-04-15 16:26:19 -0500
(Thu, 15 Apr 2010) | 13 lines Merged revisions 257544 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r257544 | tilghman | 2010-04-15 16:23:24 -0500 (Thu, 15 Apr 2010)
| 6 lines Allow application options with arguments to contain
parentheses, through a variety of escaping techniques. Fixes
SWP-1194 (ABE-2143). Review:
https://reviewboard.asterisk.org/r/604/ ........ ................
* /, channels/chan_sip.c: Merged revisions 257493 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r257493 | tilghman | 2010-04-15 15:30:15 -0500 (Thu, 15 Apr 2010)
| 20 lines Merged revisions 257467 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r257467 | tilghman | 2010-04-15 15:24:50 -0500 (Thu, 15 Apr 2010)
| 13 lines Don't recreate peer, when responding to a repeated
deregistration attempt. When a reply to a deregistration is lost
in transmit, the client retries the deregistration. Previously,
this would cause a realtime/autocreate peer to be loaded back
into memory, after it had already been correctly purged. Instead,
we just want to resend the reply without loading the peer.
(closes issue #16908) Reported by: kkm Patches:
20100412__issue16908.diff.txt uploaded by tilghman (license 14)
Tested by: kkm ........ ................
2010-04-15 19:42 +0000 [r257344-257428] Leif Madsen <lmadsen@digium.com>
* doc/backtrace.txt: Merged revisions 257427 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r257427 | lmadsen | 2010-04-15 14:41:05 -0500 (Thu, 15 Apr 2010)
| 21 lines Merged revisions 257426 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r257426 | lmadsen | 2010-04-15 14:40:33 -0500 (Thu, 15 Apr 2010)
| 13 lines Update backtrace.txt documentation. Update the
backtrace.txt documentation so it conforms to the same layout as
other documents we've been working on recently. Additionally, add
a bunch of new information about gathering backtraces for crashes
and deadlocks, along with ways of verifying your file before
uploading it. Create a couple of one line commands for people to
generate the files we need. (closes issue #17190) Reported by:
lmadsen Patches: backtrace.txt.patch-2 uploaded by lmadsen
(license 10) Tested by: lmadsen, pabelanger ........
................
* doc/backtrace.txt: Merged revisions 257343 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r257343 | lmadsen | 2010-04-15 08:44:38 -0500 (Thu, 15 Apr 2010)
| 9 lines Merged revisions 257342 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r257342 | lmadsen | 2010-04-15 08:41:45 -0500 (Thu, 15 Apr 2010)
| 1 line Update address of the bug tracker. ........
................
2010-04-14 23:00 +0000 [r257265] Tilghman Lesher <tlesher@digium.com>
* configs/features.conf.sample, /, main/features.c: Merged
revisions 257262 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r257262 |
tilghman | 2010-04-14 17:57:35 -0500 (Wed, 14 Apr 2010) | 15
lines Yet another issue where the conversion of the application
delimiter to comma caused an issue. Application arguments within
the feature map could possibly contain a comma, which conflicts
with the syntax of the features.conf configuration file. This
patch allows the argument to be wrapped in parentheses or quoted,
to allow the application arguments to be interpreted as a single
configuration parameter. (closes issue #16646) Reported by:
pinga-fogo Patches: 20100414__issue16646.diff.txt uploaded by
tilghman (license 14) Tested by: tilghman Review:
https://reviewboard.asterisk.org/r/547/ ........
2010-04-13 19:20 +0000 [r257210] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c: Merged revisions 257191 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r257191 |
tilghman | 2010-04-13 14:17:48 -0500 (Tue, 13 Apr 2010) | 10
lines Also unref the pvt when we delete the provisional keepalive
job. (closes issue #16774) Reported by: kowalma Patches:
20100315__issue16774.diff.txt uploaded by tilghman (license 14)
Tested by: falves11, jamicque Review:
https://reviewboard.asterisk.org/r/591/ ........
2010-04-13 18:43 +0000 [r257184] Matthew Nicholson <mnicholson@digium.com>
* main/manager.c, /, configs/manager.conf.sample: Merged revisions
257146 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r257146 | mnicholson | 2010-04-13 13:10:30 -0500 (Tue, 13 Apr
2010) | 16 lines Merged revisions 257070 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r257070 | mnicholson | 2010-04-13 11:46:30 -0500 (Tue, 13 Apr
2010) | 9 lines Add an option to restore past broken behavor of
the Events manager action Before r238915, certain values for the
EventMask parameter of the Events action would result in no
response being returned. This patch adds an option to restore
that broken behavior. Also while fixing this bug I discovered
that passing an empty EventMasks parameter would also result in
no response being returned, this has been fixed as well while
being preserved when the broken behavior is requested. (closes
issue #17023) Reported by: nblasgen Review:
https://reviewboard.asterisk.org/r/602/ ........ ................
2010-04-13 16:38 +0000 [r257068] Tilghman Lesher <tlesher@digium.com>
* cdr/cdr_sqlite3_custom.c, /: Merged revisions 257065 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r257065 | tilghman | 2010-04-13 11:33:21 -0500 (Tue, 13 Apr 2010)
| 8 lines Ensure that we can have commas within cdr values.
(closes issue #17001) Reported by: snuffy Patches:
20100412__issue17001.diff.txt uploaded by tilghman (license 14)
Tested by: snuffy ........
2010-04-12 17:30 +0000 [r256822-256902] Leif Madsen <lmadsen@digium.com>
* doc/HOWTO_collect_debug_information.txt (added): Merged revisions
256901 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r256901 | lmadsen | 2010-04-12 12:29:53 -0500 (Mon, 12 Apr 2010)
| 23 lines Merged revisions 256900 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r256900 | lmadsen | 2010-04-12 12:29:26 -0500 (Mon, 12 Apr 2010)
| 15 lines Add How-To document on collecting debugging info for
issues.asterisk.org Paul Belanger has been helping a lot with bug
tracking recently and created this document that we can now point
to when additional debugging information is required. This
document will help those filing issues to know how to get the
information required when filing their issues. This will make
things easier on the developers. Initial text and changes by
pabelanger. Tweaks and editing by myself. (closes issue #17159)
Reported by: pabelanger Patches:
HOWTO_collect_debug_information.txt.patch uploaded by lmadsen
(license 10) Tested by: tzafrir, pabelanger, lmadsen ........
................
* apps/app_voicemail.c, /: Merged revisions 256860 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r256860 | lmadsen | 2010-04-12 11:16:43 -0500 (Mon, 12 Apr 2010)
| 3 lines Remove silly debug message that is not useful. (issue
#17159) ........
* /, main/logger.c: Merged revisions 256821 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r256821 |
lmadsen | 2010-04-12 09:39:37 -0500 (Mon, 12 Apr 2010) | 8 lines
CLI command logger set level auto complete. A simple patch to
enable auto tab complete. (closes issue #17152) Reported by:
pabelanger Patches: 0017152.patch uploaded by pabelanger (license
224) ........
2010-04-08 22:03 +0000 [r256483] Tilghman Lesher <tlesher@digium.com>
* main/app.c: Backport /proc/%d/fd method of closing file
descriptors to 1.6.2.
2010-04-06 19:40 +0000 [r256373] Tilghman Lesher <tlesher@digium.com>
* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
include/asterisk/lock.h: Merged revisions 256370 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r256370 | tilghman | 2010-04-06 14:28:42 -0500 (Tue, 06 Apr 2010)
| 2 lines Mac OS X does not support comparing a mutex to its
initializer. Create a test for this. ........
2010-04-06 18:53 +0000 [r256268-256368] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: CallerID channel DAHDI port FXS are empty
after the first call. The bug is exposed if MFC/R2 support is
built into asterisk (i.e., openr2.h is present in the include
path). Code that unconditionally clears the CallerID name and
number is included. Also fixed a malformed if test in mkintf()
added by issue 15883. Converted the if statement to a switch
statement for clarity. Regression of the issue 15883 fix. (closes
issue #16968) Reported by: grecco Patches: issue16968.patch
uploaded by rmudgett (license 664) (closes issue #16747) Reported
by: viniciusfontes
* channels/chan_dahdi.c, /: Merged revisions 256265 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r256265 | rmudgett | 2010-04-05 19:39:44 -0500
(Mon, 05 Apr 2010) | 12 lines Merged revisions 256225 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r256225 | rmudgett | 2010-04-05 19:10:16 -0500 (Mon, 05 Apr 2010)
| 5 lines DAHDI/PRI call to pri_channel_bridge() not protected by
PRI lock. SWP-1231 ABE-2163 ........ ................
2010-05-03 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.7 Released
2010-04-29 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.7-rc3 Released
2010-04-29 10:31 +0000 [r260053] David Vossel <dvossel@digium.com>
* include/asterisk/audiohook.h, main/audiohook.c: Fixes crash in
audiohook_write_list. (closes issue 0017052) Reported by: dvossel
Tested by: dvossel. (closes issue 0016196) Reported by: atis.
Review: https://reviewboard.asterisk.org/r/623/
2010-04-28 10:31 +0000 [r259899] David Vossel <dvossel@digium.com>
* channels/chan_local.c, main/channel.c: Resolves deadlocks in
chan_local. (closes issue 0017185) 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/
2010-04-13 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.7-rc2 Released
2010-04-13 [r257210] Tilghman Lesher <tlesher@digium.com>
Also unref the pvt when we delete the provisional keepalive job.
(closes issue #16774)
Reported by: kowalma
Patches:
20100315__issue16774.diff.txt uploaded by tilghman (license 14)
Tested by: falves11, jamicque
Review: https://reviewboard.asterisk.org/r/591/
2010-04-05 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.7-rc1 Released
2010-04-05 15:15 +0000 [r256162] Leif Madsen <lmadsen@digium.com>
* doc/tex/localchannel.tex, /: Merged revisions 256161 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r256161 | lmadsen | 2010-04-05 10:14:53 -0500 (Mon, 05 Apr 2010)
| 1 line Fix for localchannel.tex to allow PDFs to be generated
again. ........
2010-04-02 23:56 +0000 [r256013-256020] Russell Bryant <russell@digium.com>
* /, apps/app_meetme.c: Merged revisions 256019 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r256019 |
russell | 2010-04-02 18:55:57 -0500 (Fri, 02 Apr 2010) | 10 lines
Export MEETMEBOOKID and fix pin-less conferences with realtime
conferences (closes issue #16866) Reported by: DEA Patches:
rt-meetme-options.txt uploaded by DEA (license 3) Tested by: DEA
Review: https://reviewboard.asterisk.org/r/582/ ........
* channels/chan_local.c, /: Merged revisions 256015 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r256015 | russell | 2010-04-02 18:46:45 -0500
(Fri, 02 Apr 2010) | 16 lines Merged revisions 256014 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r256014 | russell | 2010-04-02 18:45:56 -0500 (Fri, 02 Apr 2010)
| 9 lines Resolve a deadlock that occurs due to a pointless call
to ast_bridged_channel() (closes issue #16840) Reported by:
bzing2 Patches: patch.txt uploaded by bzing2 (license 902)
issue_16840.rev1.diff uploaded by russell (license 2) Tested by:
bzing2, russell ........ ................
* main/channel.c, /: Merged revisions 256010 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r256010 | russell | 2010-04-02 18:30:58 -0500 (Fri, 02 Apr 2010)
| 9 lines Merged revisions 256009 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r256009 | russell | 2010-04-02 18:30:15 -0500 (Fri, 02 Apr 2010)
| 2 lines Remove extremely verbose debug message. ........
................
2010-04-02 20:20 +0000 [r255955] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c, /: Merged revisions 255952 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r255952 |
tilghman | 2010-04-02 15:19:01 -0500 (Fri, 02 Apr 2010) | 8 lines
Pass the PID of the Asterisk process, not the PID of the canary.
(closes issue #17065) Reported by: globalnetinc Patches:
astcanary.patch uploaded by makoto (license 38) Tested by: frawd,
globalnetinc ........
2010-04-01 18:21 +0000 [r255676-255816] Tilghman Lesher <tlesher@digium.com>
* /, include/asterisk/lock.h: Merged revisions 255796 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r255796 | tilghman | 2010-04-01 13:16:37 -0500 (Thu, 01 Apr 2010)
| 7 lines Fix DEBUG_THREADS build on Darwin. (closes issue
#16828) Reported by: oej Patches: 20100331__issue16828.diff.txt
uploaded by tilghman (license 14) ........
* apps/app_voicemail.c, /: Recorded merge of revisions 255592 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r255592 | tilghman | 2010-03-31 14:13:02 -0500
(Wed, 31 Mar 2010) | 22 lines Recorded merge of revisions 255591
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r255591 | tilghman | 2010-03-31 14:09:46 -0500 (Wed, 31 Mar 2010)
| 15 lines Ensure line terminators in email are consistent. Fixes
an issue with certain Mail Transport Agents, where attachments
are not interpreted correctly. (closes issue #16557) Reported by:
jcovert Patches: 20100308__issue16557__1.4.diff.txt uploaded by
tilghman (license 14) 20100308__issue16557__1.6.0.diff.txt
uploaded by tilghman (license 14)
20100308__issue16557__trunk.diff.txt uploaded by tilghman
(license 14) Tested by: ebroad, zktech Reviewboard:
https://reviewboard.asterisk.org/r/544/ ........ ................
2010-03-31 17:49 +0000 [r255505] Leif Madsen <lmadsen@digium.com>
* configs/sip.conf.sample, apps/app_dial.c: Merged revisions 255504
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r255504 | lmadsen | 2010-03-31 12:48:09 -0500 (Wed, 31
Mar 2010) | 5 lines Add documentation clarifying when 't' and 'T'
can be used. (closes issue #17021) Reported by: kovzol Tested by:
lmadsen, kovzol, davidw, ebroad ........
2010-03-30 20:58 +0000 [r255326-255413] Russell Bryant <russell@digium.com>
* /, channels/chan_h323.c: Merged revisions 255410 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r255410 | russell | 2010-03-30 15:56:26 -0500
(Tue, 30 Mar 2010) | 9 lines Merged revisions 255409 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r255409 | russell | 2010-03-30 15:56:00 -0500 (Tue, 30
Mar 2010) | 2 lines Don't kill Asterisk if the H323 listener does
not start. ........ ................
* /, pbx/pbx_dundi.c: Merged revisions 255323 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r255323 | russell | 2010-03-30 11:07:49 -0500 (Tue, 30 Mar 2010)
| 9 lines Merged revisions 255322 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r255322 | russell | 2010-03-30 11:06:06 -0500 (Tue, 30 Mar 2010)
| 2 lines Don't make Asterisk not start if pbx_dundi fails to
initialize. ........ ................
2010-03-26 19:28 +0000 [r255023-255067] Leif Madsen <lmadsen@digium.com>
* configs/sip.conf.sample, /: Merged revisions 255066 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r255066 | lmadsen | 2010-03-26 14:27:56 -0500 (Fri, 26 Mar 2010)
| 6 lines Replace some documentation from 1.6.x back into trunk.
This documentation associated wth tlsbindaddr is still useful so
lets synchronize it between trunk and 1.6.x branches. (issue
#17054) ........
* configs/sip.conf.sample, /: Merged revisions 255021 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r255021 | lmadsen | 2010-03-26 14:07:38 -0500 (Fri, 26 Mar 2010)
| 8 lines Update confusing documentation for tlsbindaddr. Update
some confusing documentation for the tlsbindaddr option in
sip.conf.sample. Point at a link instead which has better
documentation. (closes issue #17054) Reported by: klaus3000
........
2010-03-25 20:43 +0000 [r254770-254805] Jason Parker <jparker@digium.com>
* utils/Makefile, /: Merged revisions 254802 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r254802 | qwell | 2010-03-25 15:41:49 -0500 (Thu, 25 Mar 2010) |
9 lines Merged revisions 254800 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r254800 | qwell | 2010-03-25 15:41:15 -0500 (Thu, 25 Mar 2010) |
1 line Don't remove local copies of utils in uninstall. ........
................
* main/astobj2.c, include/asterisk/astobj2.h: Fix DEBUG_THREADS
issue with out-of-tree modules. Take 2, without ABI breakage this
time. Review: https://reviewboard.asterisk.org/r/588/
2010-03-25 20:09 +0000 [r254721] Russell Bryant <russell@digium.com>
* channels/chan_usbradio.c, /: Merged revisions 254718 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r254718 | russell | 2010-03-25 15:08:40 -0500 (Thu, 25 Mar 2010)
| 2 lines chan_usbradio depends on alsa. ........
2010-03-25 17:47 +0000 [r254556] Mark Michelson <mmichelson@digium.com>
* include/asterisk/acl.h, /: Merged revisions 254553 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r254553 | mmichelson | 2010-03-25 12:42:36 -0500
(Thu, 25 Mar 2010) | 11 lines Merged revisions 254552 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r254552 | mmichelson | 2010-03-25 12:33:35 -0500 (Thu, 25 Mar
2010) | 5 lines Add doxygen for acl.h Review:
https://reviewboard.asterisk.org/r/528 ........ ................
2010-03-25 17:21 +0000 [r254548] Sean Bright <sean@malleable.com>
* channels/chan_sip.c: Initialize stream to avoid a compilation
error.
2010-03-25 17:12 +0000 [r254542] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix potential crashes from trying to
reference nonexistent RTP streams.
2010-03-25 16:26 +0000 [r254499] Terry Wilson <twilson@digium.com>
* /, main/file.c: Merged revisions 254453 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r254453 | twilson | 2010-03-25 11:03:51 -0500 (Thu, 25 Mar 2010)
| 9 lines Merged revisions 254451 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r254451 | twilson | 2010-03-25 10:57:29 -0500 (Thu, 25 Mar 2010)
| 2 lines Handle new SRCCHANGE control message here too ........
................
2010-03-25 16:22 +0000 [r254482] Mark Michelson <mmichelson@digium.com>
* main/rtp.c, /: Recorded merge of revisions 254454 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r254454 | mmichelson | 2010-03-25 11:04:48 -0500
(Thu, 25 Mar 2010) | 50 lines Recorded merge of revisions 254452
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r254452 | mmichelson | 2010-03-25 10:59:56 -0500 (Thu, 25 Mar
2010) | 44 lines Several fixes regarding RFC2833 DTMF detection.
Here is a copy and paste of the details from my request on
reviewboard that dealt with these changes: Fix 1. The first
change in place is to fix Mantis issue 15811, which deals with a
situation where Asterisk will incorrectly interpret out of order
RFC2833 frames as duplicate DTMF digits. For instance, we would
receive a sequence like: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3:
DTMF 1 seqno 4: DTMF 1 seqno 6: DTMF 1 (end) seqno 5: DTMF 1
seqno 7: DTMF 1 (end) seqno 8: DTMF 1 (end) Prior to this patch
when we received the frame with seqno 5, we would interpret this
as a new DTMF 1. With this patch, we will check the seqno of the
incoming digit and not process the frame if the seqno is lower
than the last recorded seqno. Note that we do not record the
seqno of the dropped DTMF frame for future processing. While the
above situation is what was designed to be fixed, the patch is
written in such a way that the following would also be fixed too:
seqno 9: DTMF 1 seqno 10: DTMF 1 (end) seqno 11: DTMF 1 (end)
seqno 13: DTMF 2 seqno 12: DTMF 1 (end) seqno 14: DTMF 2 seqno
15: DTMF 2 (end) seqno 16: DTMF 2 (end) seqno 17: DTMF 2 (end) In
this second situation, the beginning of the DTMF 2 arrives before
the final end frame of the DTMF 1. With the patch, seqno 12 is no
processed and thus we properly interpret the DTMF. Fix 2. The
second change in place is to fix an issue like the following:
seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 (end) *packet
lost* seqno 4: DTMF 1 (end) *packet lost* seqno 5: DTMF 1 (end)
*packet lost* seqno 6: DTMF 2 When we receive seqno 6, we had
code in place that was supposed to properly end the previously
unended DTMF 1. The problem was that the code was essentially a
no-op. The code would set up an end frame for the DTMF 1 but
would immediately overwrite the frame with the begin for DTMF 2.
I changed process_dtmf_rfc2833() so that instead of returning a
single frame, it is given as an output parameter a list of
frames. Each frame that needs to be returned is appended to this
list. Fix 3. The final change is a minor one where an
AST_CONTROL_SRCCHANGE frame could get lost. If we process a cisco
DTMF or an RFC 3389 frame and no frame was returned, then we
would return &ast_null_frame. The problem is that earlier in the
function, we may have generated an AST_CONTROL_SRCCHANGE frame
and put it in the list of frames we wish to return. This frame
would be lost in such a case. The patch fixes this problem
........ ................
2010-03-25 15:21 +0000 [r254447] Leif Madsen <lmadsen@digium.com>
* /, res/res_agi.c: Merged revisions 254446 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r254446 |
lmadsen | 2010-03-25 10:21:26 -0500 (Thu, 25 Mar 2010) | 9 lines
handle_speechset has 4 arguments. Update code to reflect that
handle_speechset has 4 arguments. (closes issue #17093) Reported
by: gpatri Patches: res_agi.patch uploaded by gpatri (license
1014) Tested by: pabelanger, mmichelson ........
2010-03-24 17:19 +0000 [r254288] Jeff Peeler <jpeeler@digium.com>
* res/res_monitor.c, /: Merged revisions 254277 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r254277 | jpeeler | 2010-03-24 12:15:05 -0500 (Wed, 24 Mar 2010)
| 78 lines Merged revisions 254235 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r254235 | jpeeler | 2010-03-23 19:37:23 -0500 (Tue, 23 Mar 2010)
| 72 lines Ensure that monitor recordings are written to the
correct location (again) This is an extension to 248860. As such
the dialplan test has been extended: ; non absolute path, not
combined exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test)
exten => 5040, n, dial(sip/5001) ; absolute path, not combined
exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2) exten =>
5041, n, dial(sip/5001) ; no path, not combined exten => 5042, 1,
monitor(wav,monitor_test3) exten => 5042, n, dial(sip/5001) ;
combined: changemonitor from non absolute to no path (leaves
tmp/jeff) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test4,m)
exten => 5043, n, changemonitor(monitor_test5) exten => 5043, n,
dial(sip/5001) ; combined: changemonitor from no path to non
absolute path exten => 5044, 1, monitor(wav,monitor_test6,m)
exten => 5044, n, changemonitor(tmp/jeff/monitor_test7) ; this
wasn't possible before exten => 5044, n, dial(sip/5001) ; non
absolute path, combined exten => 5045, 1,
monitor(wav,tmp/jeff/monitor_test8,m) exten => 5045, n,
dial(sip/5001) ; absolute path, combined exten => 5046, 1,
monitor(wav,/tmp/jeff/monitor_test9,m) exten => 5046, n,
dial(sip/5001) ; no path, combined exten => 5047, 1,
monitor(wav,monitor_test10,m) exten => 5047, n, dial(sip/5001) ;
combined: changemonitor from non absolute to absolute (leaves
tmp/jeff) exten => 5048, 1,
monitor(wav,tmp/jeff/monitor_test11,m) exten => 5048, n,
changemonitor(/tmp/jeff/monitor_test12) exten => 5048, n,
dial(sip/5001) ; combined: changemonitor from absolute to non
absolute (leaves /tmp/jeff) exten => 5049, 1,
monitor(wav,/tmp/jeff/monitor_test13,m) exten => 5049, n,
changemonitor(tmp/jeff/monitor_test14) exten => 5049, n,
dial(sip/5001) ; combined: changemonitor from no path to absolute
exten => 5050, 1, monitor(wav,monitor_test15,m) exten => 5050, n,
changemonitor(/tmp/jeff/monitor_test16) exten => 5050, n,
dial(sip/5001) ; combined: changemonitor from absolute to no path
(leaves /tmp/jeff) exten => 5051, 1,
monitor(wav,/tmp/jeff/monitor_test17,m) exten => 5051, n,
changemonitor(monitor_test18) exten => 5051, n, dial(sip/5001) ;
not combined: changemonitor from non absolute to no path (leaves
tmp/jeff) exten => 5052, 1, monitor(wav,tmp/jeff/monitor_test19)
exten => 5052, n, changemonitor(monitor_test20) exten => 5052, n,
dial(sip/5001) ; not combined: changemonitor from no path to non
absolute exten => 5053, 1, monitor(wav,monitor_test21) exten =>
5053, n, changemonitor(tmp/jeff/monitor_test22) exten => 5053, n,
dial(sip/5001) ; not combined: changemonitor from non absolute to
absolute (leaves tmp/jeff) exten => 5054, 1,
monitor(wav,tmp/jeff/monitor_test23) exten => 5054, n,
changemonitor(/tmp/jeff/monitor_test24) exten => 5054, n,
dial(sip/5001) ; not combined: changemonitor from absolute to non
absolute (leaves /tmp/jeff) exten => 5055, 1,
monitor(wav,/tmp/jeff/monitor_test24) exten => 5055, n,
changemonitor(tmp/jeff/monitor_test25) exten => 5055, n,
dial(sip/5001) ; not combined: changemonitor from no path to
absolute exten => 5056, 1, monitor(wav,monitor_test26) exten =>
5056, n, changemonitor(/tmp/jeff/monitor_test27) exten => 5056,
n, dial(sip/5001) ; not combined: changemonitor from absolute to
no path (leaves /tmp/jeff) exten => 5057, 1,
monitor(wav,/tmp/jeff/monitor_test28) exten => 5057, n,
changemonitor(monitor_test29) exten => 5057, n, dial(sip/5001)
........ ................
2010-03-23 22:05 +0000 [r254131] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* tests/Makefile, /: Merged revisions 254001 via svnmerge from
http://svn.digium.com/svn/asterisk/trunk ........ r254001 |
tzafrir | 2010-03-23 21:19:52 +0200 (Tue, 23 Mar 2010) | 2 lines
Change the name of the category 'TEST' to match the name of the
subdir ........
2010-03-23 21:20 +0000 [r254068] Jeff Peeler <jpeeler@digium.com>
* main/channel.c, /: Merged revisions 254050 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r254050 |
jpeeler | 2010-03-23 16:17:23 -0500 (Tue, 23 Mar 2010) | 14 lines
Exit native bridging early for greater timing accuracy with
warnings This changes native bridging to break one millisecond
early so that the more accurate timeval calculations done in the
generic bridge can be performed using the bridge config.
Currently the time between exiting native bridging slightly late
can sometimes cause a large enough discrepancy for warnings to be
missed. For the record, 1.4 does not attempt to native bridge at
all when warnings are enabled. (closes issue #15815) Reported by:
adomjan Review: https://reviewboard.asterisk.org/r/577/ ........
2010-03-22 19:55 +0000 [r253801] Matthew Nicholson <mnicholson@digium.com>
* /, main/features.c: Merged revisions 253800 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r253800 | mnicholson | 2010-03-22 14:52:52 -0500 (Mon, 22 Mar
2010) | 11 lines Merged revisions 253799 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r253799 | mnicholson | 2010-03-22 14:50:00 -0500 (Mon, 22 Mar
2010) | 4 lines Unconditionally copy the caller's account code to
the called party. (related to issue #16331) ........
................
2010-03-22 19:06 +0000 [r253714-253760] Tilghman Lesher <tlesher@digium.com>
* /, contrib/scripts/dbsep.cgi: Merged revisions 253758 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r253758 | tilghman | 2010-03-22 14:05:27 -0500 (Mon, 22
Mar 2010) | 2 lines Update query should be an UPDATE, not a
SELECT. ........
* /, contrib/scripts/dbsep.cgi: Merged revisions 253755 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r253755 | tilghman | 2010-03-22 13:58:48 -0500 (Mon, 22
Mar 2010) | 4 lines Return the list for later manipulation. This
fixes an issue with the update procedure. Debugging with
mmichelson. ........
* configs/dbsep.conf.sample, /, contrib/scripts/dbsep.cgi: Merged
revisions 253712 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r253712 |
tilghman | 2010-03-22 11:59:35 -0500 (Mon, 22 Mar 2010) | 2 lines
Accomodate equal signs in DSNs and add documentation, based upon
mmichelson's feedback. ........
2010-03-20 17:33 +0000 [r253595-253620] Russell Bryant <russell@digium.com>
* cdr/cdr_pgsql.c, main/stdtime/localtime.c, main/tcptls.c, /,
main/features.c: Merged revisions 253540 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r253540 |
russell | 2010-03-20 07:03:07 -0500 (Sat, 20 Mar 2010) | 2 lines
Resolve more compiler warnings on FreeBSD. ........
* apps/app_followme.c, apps/app_dial.c, /: Merged revisions 253538
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r253538 | russell | 2010-03-20 06:43:08 -0500 (Sat, 20
Mar 2010) | 2 lines Resolve compiler warnings on FreeBSD.
........
* /, pbx/pbx_dundi.c: Merged revisions 253537 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r253537 |
russell | 2010-03-20 06:39:39 -0500 (Sat, 20 Mar 2010) | 2 lines
Resolve a compiler warning on FreeBSD. ........
* channels/chan_dahdi.c, /: Merged revisions 253536 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r253536 | russell | 2010-03-20 06:33:30 -0500 (Sat, 20 Mar 2010)
| 4 lines Use SHRT_MAX instead of MAXSHORT. These changes fix
build issues I had with this module on FreeBSD. ........
2010-03-19 08:05 +0000 [r253492] Alec L Davis <sivad.a@paradise.net.nz>
* main/astobj2.c, /: Merged revisions 253490 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r253490 |
alecdavis | 2010-03-19 20:37:00 +1300 (Fri, 19 Mar 2010) | 19
lines prevent segfault if bad magic number is encountered.
internal_ao2_ref uses INTERNAL_OBJ which mzy report 'bad magic
number', but internal_ao2_ref continues on, causing segfault.
Although AO2_MAGIC number is checked by INTERNAL_OBJ before
internal_ao2_ref is called, A02_MAGIC is being destroyed (or a
wrong pointer) by the time internal_ao2_ref uses INTERNAL_OBJ.
internal_ao2_ref now returns -1 if INTERNAL_OBJ encouters a bad
magic number. (issue #17037) Reported by: alecdavis Patches:
bug17037.diff.txt uploaded by alecdavis (license 585) Tested by:
alecdavis ........
2010-03-18 17:54 +0000 [r253257-253346] Leif Madsen <lmadsen@digium.com>
* /, apps/app_userevent.c: Merged revisions 253345 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r253345 | lmadsen | 2010-03-18 12:52:35 -0500 (Thu, 18 Mar 2010)
| 7 lines Change usage of pipe to comma in UserEvent docs. Change
the example usage of pipe as a separator to comma in the
UserEvent documentation. (closes issue #16961) Reported by:
jlpedrosa ........
* doc/tex/localchannel.tex: Merged revisions 253256 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r253256 | lmadsen | 2010-03-18 10:46:52 -0500 (Thu, 18 Mar 2010)
| 9 lines Update to new Local channel documentation. Add same
changes as commit to 1.4, but convert to TeX. (issue #16963)
Reported by: kobaz Patches: localchannel-2.txt uploaded by kobaz
(license 834) ........
2010-03-17 16:25 +0000 [r253158] Terry Wilson <twilson@digium.com>
* main/rtp.c, channels/chan_skinny.c, channels/chan_h323.c,
channels/chan_mgcp.c, channels/chan_sip.c,
include/asterisk/rtp.h: Revert API change in release branches
This re-renames ast_rtp_update_source to ast_rtp_new_source
2010-03-17 00:41 +0000 [r253029-253033] Leif Madsen <lmadsen@digium.com>
* main/xmldoc.c, /: Merged revisions 253032 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r253032 |
lmadsen | 2010-03-16 19:40:51 -0500 (Tue, 16 Mar 2010) | 1 line
Fix a typo. ........
* configs/say.conf.sample, /: Merged revisions 253028 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r253028 | lmadsen | 2010-03-16 19:29:06 -0500
(Tue, 16 Mar 2010) | 13 lines Merged revisions 253018 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r253018 | lmadsen | 2010-03-16 19:26:19 -0500 (Tue, 16 Mar 2010)
| 6 lines Add french snipset to say.conf. Add the french snipset
to say.conf. (Closes issue #15799) ........ ................
2010-03-16 23:54 +0000 [r252978] Tilghman Lesher <tlesher@digium.com>
* apps/app_stack.c, /: Merged revisions 252976 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r252976 |
tilghman | 2010-03-16 18:49:35 -0500 (Tue, 16 Mar 2010) | 8 lines
Mask out previous arguments on each nested invocation of Gosub.
(closes issue #16758) Reported by: wdoekes Patches:
20100316__issue16758.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/561/ ........
2010-03-16 19:38 +0000 [r252850] Sean Bright <sean@malleable.com>
* res/res_clialiases.c, /: Merged revisions 252848 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r252848 | seanbright | 2010-03-16 15:36:24 -0400 (Tue, 16 Mar
2010) | 10 lines Include an extra newline after "Aliased CLI
command" to get back the prompt. The other issue mentioned in
this bug will be more difficult to resolve since we have no idea
(right now) of knowing if the command that is aliased has been
installed yet. (issue #16978) Reported by: jw-asterisk Tested by:
seanbright ........
2010-03-16 19:02 +0000 [r252770] Russell Bryant <russell@digium.com>
* utils/Makefile, /: Merged revisions 252767 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r252767 | russell | 2010-03-16 14:01:04 -0500 (Tue, 16 Mar 2010)
| 13 lines Merged revisions 252766 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r252766 | russell | 2010-03-16 14:00:43 -0500 (Tue, 16 Mar 2010)
| 6 lines Don't treat warnings as errors for muted. muted
supports OS X, but uses functions marked as deprecated in 10.6.
However, the functions are still supported, so just ignore the
warnings for now and allow the build to proceed. ........
................
2010-03-16 18:49 +0000 [r252763] Leif Madsen <lmadsen@digium.com>
* configs/extensions.ael.sample, /: Merged revisions 252762 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r252762 | lmadsen | 2010-03-16 13:48:22 -0500
(Tue, 16 Mar 2010) | 15 lines Merged revisions 252761 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r252761 | lmadsen | 2010-03-16 13:46:20 -0500 (Tue, 16 Mar 2010)
| 7 lines Additional extensions.ael global variable fixes. Fixing
up a couple more overlapping global variable namespaces shared
with extensions.conf.sample. Also noticed a few of the lines that
were commented out didn't have the closing semi-colon so I added
that as well. (issue #17035) ........ ................
2010-03-15 21:59 +0000 [r252626] Sean Bright <sean@malleable.com>
* /, apps/app_meetme.c: Merged revisions 252623 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r252623 |
seanbright | 2010-03-15 17:55:44 -0400 (Mon, 15 Mar 2010) | 4
lines Resolve a crash in SLATrunk when the specified trunk
doesn't exist. Reported by philipp64 in #asterisk-dev. ........
2010-03-15 21:54 +0000 [r252622] Tilghman Lesher <tlesher@digium.com>
* contrib/init.d/org.asterisk.asterisk.plist, /: Merged revisions
252619 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r252619 | tilghman | 2010-03-15 16:51:55 -0500 (Mon, 15 Mar 2010)
| 9 lines Merged revisions 252617 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r252617 | tilghman | 2010-03-15 16:43:14 -0500 (Mon, 15 Mar 2010)
| 2 lines Uh, yeah. Umask. I'm stupid. ........ ................
2010-03-15 20:53 +0000 [r252535] Leif Madsen <lmadsen@digium.com>
* configs/extensions.ael.sample: Merged revisions 252534 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r252534 | lmadsen | 2010-03-15 15:52:32 -0500
(Mon, 15 Mar 2010) | 15 lines Merged revisions 252533 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r252533 | lmadsen | 2010-03-15 15:48:56 -0500 (Mon, 15 Mar 2010)
| 7 lines Update extensions.ael file to not overlap
extensions.conf. Updated the extensions.ael file so the global
variables don't overlap those that we have in extensions.conf
(sample files). This way unexpected things won't happed hopefully
if both pbx_ael and res_config are loaded. (closes issue #17035)
Reported by: pprindeville ........ ................
2010-03-15 05:04 +0000 [r252365-252444] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c: Merged revisions 252442 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r252442 |
tilghman | 2010-03-14 23:25:35 -0500 (Sun, 14 Mar 2010) | 7 lines
THIS IS NOT PYTHON. Indentation doesn't matter, only braces do.
(closes issue #17025) Reported by: smurfix Patches: sip.patch
uploaded by smurfix (license 547) ........
* main/asterisk.c, Makefile,
contrib/init.d/org.asterisk.asterisk.plist (added), /: Merged
revisions 252362 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r252362 | tilghman | 2010-03-14 20:37:04 -0500 (Sun, 14 Mar 2010)
| 11 lines Merged revisions 252361 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r252361 | tilghman | 2010-03-14 20:33:50 -0500 (Sun, 14 Mar 2010)
| 4 lines Launch Asterisk on Mac OS X with launchd. Reviewboard:
https://reviewboard.asterisk.org/r/551/ ........ ................
2010-03-14 17:48 +0000 [r252317] Sean Bright <sean@malleable.com>
* cdr/cdr_sqlite3_custom.c, /: Merged revisions 252314 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r252314 | seanbright | 2010-03-14 13:43:46 -0400 (Sun, 14 Mar
2010) | 8 lines Fix building CDR and CEL SQLite3 modules. They
added a sqlite3_log() function which was conflicting with our
function names. (closes issue #17017) Reported by: alephlg
........
2010-03-13 00:32 +0000 [r252137-252178] Terry Wilson <twilson@digium.com>
* main/rtp.c: Remove unusued field
* configs/sip.conf.sample, include/asterisk/frame.h, main/rtp.c,
channels/chan_mgcp.c, main/channel.c, /, channels/chan_sip.c,
channels/chan_skinny.c, include/asterisk/rtp.h,
channels/chan_h323.c: Merged revisions 252089 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r252089 |
twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines
Only change the RTP ssrc when we see that it has changed This
change basically reverts the change reviewed in
https://reviewboard.asterisk.org/r/374/ and instead limits the
updating of the RTP synchronization source to only those times
when we detect that the other side of the conversation has
changed the ssrc. The problem is that SRCUPDATE control frames
are sent many times where we don't want a new ssrc, including
whenever Asterisk has to send DTMF in a normal bridge. This is
also not the first time that this mistake has been made. The
initial implementation of the ast_rtp_new_source function also
changed the ssrc--and then it was removed because of this same
issue. Then, we put it back in again to fix a different issue.
This patch attempts to only change the ssrc when we see that the
other side of the conversation has changed the ssrc. It also
renames some functions to make their purpose more clear. Review:
https://reviewboard.asterisk.org/r/540/ ........
2010-03-12 22:05 +0000 [r252090] Moises Silva <moises.silva@gmail.com>
* channels/chan_dahdi.c, /: Merged revisions 252088 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r252088 | moy | 2010-03-12 16:57:40 -0500 (Fri, 12 Mar 2010) | 1
line add missing mfcr2_skip_category setting ........
2010-03-12 19:50 +0000 [r251994] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, /: Merged revisions 251989 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r251989 | tilghman | 2010-03-12 13:43:23 -0600 (Fri, 12 Mar 2010)
| 8 lines Don't override a user option with the global option.
(closes issue #16849) Reported by: ip-rob Patches:
20100311__issue16849.diff.txt uploaded by tilghman (license 14)
Tested by: ip-rob ........
2010-03-12 19:49 +0000 [r251991] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 251946 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r251946 | rmudgett | 2010-03-12 13:05:40 -0600 (Fri, 12 Mar 2010)
| 1 line Doxegen this chan_dahdi lock. ........
2010-03-11 21:08 +0000 [r251879-251887] Tilghman Lesher <tlesher@digium.com>
* apps/app_exec.c, /: Merged revisions 251884 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r251884 |
tilghman | 2010-03-11 15:07:07 -0600 (Thu, 11 Mar 2010) | 8 lines
Because ExecIf needs to reprocess arguments, it's best if we
don't remove quotes during parsing. (closes issue #16905)
Reported by: ip-rob Patches: 20100303__issue16905.diff.txt
uploaded by tilghman (license 14) Tested by: ip-rob ........
* apps/app_system.c, /: Merged revisions 251877 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r251877 |
tilghman | 2010-03-11 14:25:02 -0600 (Thu, 11 Mar 2010) | 8 lines
If the argument to the system application is quoted, ensure we
remove the quotes before trying to execute. (closes issue #16842)
Reported by: ip-rob Patches: 20100310__issue16842.diff.txt
uploaded by tilghman (license 14) Tested by: ip-rob ........
2010-03-11 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.6 released
2010-03-05 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.6-rc2 released
2010-03-05 Tilghman Lesher <tlesher@digium.com>
* /, apps/app_voicemail.c: Merged revisions 250913 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r250913 | tilghman
| 2010-03-04 22:37:36 -0600 (Thu, 04 Mar 2010) | 7 lines Missing quote in
ODBC query. (closes issue #16953) Reported by: elguero Patches:
app_voicemail-odbc-syntax-fix.diff uploaded by elguero (license 37)
........
2010-03-04 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.6-rc1 released
2010-03-03 21:24 +0000 [r250610] Leif Madsen <lmadsen@digium.com>
* doc/tex/localchannel.tex, /: Merged revisions 250609 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r250609 | lmadsen | 2010-03-03 16:22:55 -0500 (Wed, 03 Mar 2010)
| 11 lines Update existing Local channel documentation. A
complete re-write of the Local channel documentation has been
performed, with the existing information from localchannel.txt
and localchannel.tex merged in. (closes issue #16637) Reported
by: kobaz Patches: localchannel.tex uploaded by lmadsen (license
10) localchannel.txt uploaded by lmadsen (license 10) Tested by:
lmadsen, jsmith, mmichelson ........
2010-03-03 19:13 +0000 [r250484] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 250481 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r250481 | jpeeler | 2010-03-03 13:06:06 -0600
(Wed, 03 Mar 2010) | 22 lines Merged revisions 250480 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r250480 | jpeeler | 2010-03-03 13:04:11 -0600 (Wed, 03 Mar 2010)
| 15 lines Make sure to clear red alarm after polarity reversal.
From the issue: The automatic overnight line tests (or manual
ones) used on UK (BT) lines causes a red alarm on a dahdi /
TDM400P connected channel. This is because the line uses voltage
tests (battery loss) and polarity reversal. The polarity reversal
causes chan_dahdi to initiate v23 CallerID processing but during
this the event DAHDI_EVENT_NOALARM is ignored so that the alarm
is never cleared. (closes issue #14163) Reported by: jedi98
Patches: chan_dahdi-1.4-inalarm.diff uploaded by jedi98 (license
653) Tested by: mattbrown, Chainsaw, mikeeccleston ........
................
2010-03-03 18:05 +0000 [r250253-250396] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c, /: Merged revisions 250395 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r250395 | dvossel | 2010-03-03 12:03:19 -0600
(Wed, 03 Mar 2010) | 22 lines Merged revisions 250394 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r250394 | dvossel | 2010-03-03 12:02:27 -0600 (Wed, 03 Mar 2010)
| 16 lines fixes problem with duplicate TXREQ packets When
Asterisk receives an IAX2 TXREQ packet, try_transfer() will call
store_by_transfercallno() to link the chan_iax2_pvt struct into
iax_transfercallno_pvts. If a duplicate TXREQ packet is received
for the same call, the pvt struct will be linked into
iax_transfercallno_pvts multiple times. This patch fixes this.
Thanks rain for debugging this and providing a patch! (closes
issue #16904) Reported by: rain Patches:
iax2-double-txreq-fix.diff uploaded by rain (license 327) Tested
by: rain, dvossel ........ ................
* /, channels/chan_sip.c: Merged revisions 250246 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r250246 |
dvossel | 2010-03-02 18:18:28 -0600 (Tue, 02 Mar 2010) | 2 lines
fixes signed to unsigned int comparision issue for FaxMaxDatagram
value. ........
2010-03-02 21:10 +0000 [r249953-250052] Leif Madsen <lmadsen@digium.com>
* doc/tex/imapstorage.tex, /: Merged revisions 250051 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r250051 | lmadsen | 2010-03-02 16:09:27 -0500 (Tue, 02 Mar 2010)
| 8 lines Update IMAP documentation. Update the IMAP
documentation to make it clear that storing voicemails in the
same folder as a large number of emails could potentially cause
significant slow downs when writing or retrieving voicemails.
(issue #16704) Reported by: TimeHider Tested by: lmadsen,
TimeHider ........
* configs/cdr.conf.sample: Merged revisions 250045 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r250045 | lmadsen | 2010-03-02 15:52:19 -0500
(Tue, 02 Mar 2010) | 15 lines Merged revisions 250043 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r250043 | lmadsen | 2010-03-02 15:51:35 -0500 (Tue, 02 Mar 2010)
| 7 lines Update documentation to clarify purpose of unanswered
option. (closes issue #16267) Reported by: elsto Patches:
cdr.conf.sample.patch.txt uploaded by lmadsen (license 10) Tested
by: davidw, elsto ........ ................
* doc/tex/configuration.tex, /: Merged revisions 250037 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r250037 | lmadsen | 2010-03-02 15:36:10 -0500 (Tue, 02
Mar 2010) | 4 lines Update documentation to not imply we support
overriding options. (closes issue #16855) Reported by: davidw
........
* apps/app_directory.c, /: Merged revisions 249950 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r249950 | lmadsen | 2010-03-02 14:49:48 -0500 (Tue, 02 Mar 2010)
| 4 lines Fix literal values wrapped in documentation. (closes
issue #16145) Reported by: tilghman ........
2010-03-02 19:50 +0000 [r249952] Alec L Davis <sivad.a@paradise.net.nz>
* UPGRADE-1.6.txt, main/editline/makelist.in, apps/app_echo.c,
UPGRADE.txt: revert ability to exit echo app caused a regression,
as only supported VOICE, not VIDEO etc. (issue #16880)
2010-03-02 19:26 +0000 [r249916-249933] Leif Madsen <lmadsen@digium.com>
* /, main/features.c: Merged revisions 249925 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r249925 |
lmadsen | 2010-03-02 14:24:43 -0500 (Tue, 02 Mar 2010) | 6 lines
Add missing description of the PARKINGLOT variable in XML
documentation. (closes issue #16743) Reported by: snuffy Patches:
parkingdoc.diff uploaded by snuffy (license 35) ........
* /, pbx/pbx_dundi.c: Merged revisions 249912 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r249912 |
lmadsen | 2010-03-02 14:21:19 -0500 (Tue, 02 Mar 2010) | 6 lines
Convert some DUNDI functions to XML documentation. (closes issue
#16798) Reported by: snuffy Patches: xml_dundi.diff uploaded by
snuffy (license 35) ........
2010-03-02 19:12 +0000 [r249895] David Vossel <dvossel@digium.com>
* channels/chan_console.c, channels/chan_gtalk.c,
channels/chan_oss.c, channels/misdn_config.c,
include/asterisk/abstract_jb.h, configs/alsa.conf.sample,
channels/chan_jingle.c, channels/chan_usbradio.c,
channels/chan_dahdi.c, channels/chan_skinny.c,
configs/mgcp.conf.sample, main/abstract_jb.c,
channels/chan_h323.c, channels/chan_alsa.c,
configs/sip.conf.sample, channels/chan_mgcp.c,
channels/chan_unistim.c, configs/console.conf.sample,
configs/chan_dahdi.conf.sample, channels/chan_local.c,
configs/oss.conf.sample, channels/chan_sip.c, /,
configs/usbradio.conf.sample, configs/misdn.conf.sample: Merged
revisions 249893 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r249893 |
dvossel | 2010-03-02 13:08:38 -0600 (Tue, 02 Mar 2010) | 11 lines
fixes adaptive jitterbuffer configuration When configuring the
adaptive jitterbuffer, the target_extra value not only could not
be set from the configuration, but was not even being set to its
proper default. This value is required in order for the adaptive
jitterbuffer to work correctly. To resolve this a config option
has been added to expose this value to the conf files, and a
default value is provided when no config specific value is
present. ........
2010-03-02 19:09 +0000 [r249894] Leif Madsen <lmadsen@digium.com>
* /, apps/app_confbridge.c: Merged revisions 249892 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r249892 | lmadsen | 2010-03-02 14:02:56 -0500 (Tue, 02 Mar 2010)
| 1 line Fix several XML documentation validate errors. ........
2010-03-02 09:05 +0000 [r249844] Alec L Davis <sivad.a@paradise.net.nz>
* apps/app_echo.c: fixes ability to exit echo app when called from
a ISDN channel, null frames prevent '#' exit. Now only echo back
VOICE and DTMF frames (issue #16880) Reported by: alecdavis
Patches: echo_exit_1-6-1.diff.txt uploaded by alecdavis (license
585) Tested by: alecdavis
2010-03-01 19:40 +0000 [r249675] Sean Bright <sean@malleable.com>
* apps/app_voicemail.c, /: Merged revisions 249672 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r249672 | seanbright | 2010-03-01 14:36:30 -0500
(Mon, 01 Mar 2010) | 18 lines Merged revisions 249671 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r249671 | seanbright | 2010-03-01 14:35:01 -0500 (Mon, 01 Mar
2010) | 11 lines Fix crash in app_voicemail related to message
counting. We were passing a 'struct inprocess **' and treating it
like a 'struct inprocess *' causing a segfault. (closes issue
#16921) Reported by: whardier Patches: 20100301_issue16921.patch
uploaded by seanbright (license 71) Tested by: whardier ........
................
2010-03-01 18:47 +0000 [r249625] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, /: Merged revisions 249623 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r249623 | tilghman | 2010-03-01 12:36:06 -0600 (Mon, 01 Mar 2010)
| 2 lines Constify a bit of app_voicemail, to make ODBC and IMAP
compile once again. ........
2010-03-01 17:25 +0000 [r249580] Jeff Peeler <jpeeler@digium.com>
* channels/chan_local.c, /: Merged revisions 249538 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r249538 | jpeeler | 2010-03-01 11:11:31 -0600
(Mon, 01 Mar 2010) | 18 lines Merged revisions 249536 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r249536 | jpeeler | 2010-03-01 11:02:03 -0600 (Mon, 01 Mar 2010)
| 11 lines Modify queued frames from local channels to not set
the other side to up In this case, attended transfers were broken
due to ast_feature_request_and_dial detecting the channel being
set to up before the answer frame could be read and therefore
failing to mark the channel as ready. This fix is a regression
fix for 244785, which should continue to work properly as well.
(closes issue #16816) Reported by: jamhed Tested by: jamhed,
corruptor ........ ................
2010-02-28 20:51 +0000 [r249407-249493] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, /: Merged revisions 249491 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r249491 | tilghman | 2010-02-28 14:50:01 -0600 (Sun, 28 Feb 2010)
| 5 lines Fix unit test that Alec Davis broke. (closes issue
#16927) Reported by: alecdavis ........
* apps/app_voicemail.c, include/asterisk/app.h, /: Merged revisions
249405 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r249405 |
tilghman | 2010-02-28 01:10:22 -0600 (Sun, 28 Feb 2010) | 2 lines
Properly document voicemail API documents. Also fix a crash
reported via the -dev list. ........
2010-02-27 23:04 +0000 [r249321] Alec L Davis <sivad.a@paradise.net.nz>
* channels/chan_dahdi.c: overlap receiving: automatically send CALL
PROCEEDING when dialplan starts Following Q.931 5.2.4 When the
user has determined that sufficient call information has been
received the user shall stop T302 and send CALL PROCEEDING to the
network. Previously timeouts were possible if the dialplan took a
long time to issue any response back to the network. Verified
that our local TELCO also does the same. (issue #16789) Reported
by: alecdavis Patches: overlap_receiving_trunk.diff.txt uploaded
by alecdavis (license 585) Tested by: alecdavis
2010-02-27 14:10 +0000 [r249238] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_iax2.c, /: Merged revisions 249235 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r249235 | kpfleming | 2010-02-27 09:08:35 -0500
(Sat, 27 Feb 2010) | 9 lines Merged revisions 249234 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r249234 | kpfleming | 2010-02-27 09:07:59 -0500 (Sat, 27
Feb 2010) | 1 line add a reference to the now-published IAX2 RFC
........ ................
2010-02-26 18:49 +0000 [r249190] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, /: Merged revisions 249187 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r249187 | tilghman | 2010-02-26 12:41:57 -0600 (Fri, 26 Feb 2010)
| 18 lines Cleanups to fix bugs in the VM count API functions. -
Urgent voicemails were not attached, because the attachment code
looked in the wrong folder. - Urgent voicemails were sometimes
counted twice when displaying the count of new messages. -
Backends were inconsistent as to which voicemails each API
counted. (closes issue #15654) Reported by: tomo1657 Patches:
20100225__issue15654.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman (closes issue #16448) Reported by: hevad
Review: https://reviewboard.asterisk.org/r/525/ ........
2010-02-26 17:06 +0000 [r249104] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 249101 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r249101 | mmichelson | 2010-02-26 11:04:58 -0600 (Fri, 26 Feb
2010) | 14 lines Merged revisions 249100 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r249100 | mmichelson | 2010-02-26 11:04:29 -0600 (Fri, 26 Feb
2010) | 8 lines For T.38 reINVITEs treat a 606 the same as a 488.
(closes issue #16792) Reported by: vrban Patches: t38_606.patch
uploaded by vrban (license 756) ........ ................
2010-02-25 23:12 +0000 [r248955] Jeff Peeler <jpeeler@digium.com>
* res/res_monitor.c, /: Merged revisions 248952 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r248952 | jpeeler | 2010-02-25 17:09:54 -0600 (Thu, 25 Feb 2010)
| 24 lines Merged revisions 248860 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r248860 | jpeeler | 2010-02-25 15:22:06 -0600 (Thu, 25 Feb 2010)
| 18 lines Ensure that monitor recordings are written to the
correct location (again) This is an extension to 248757. As such
the dialplan test has been extended: exten => 5040, 1,
monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n,
dial(sip/5001) exten => 5041, 1,
monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n,
dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b)
exten => 5042, n, dial(sip/5001) exten => 5043, 1,
monitor(wav,tmp/jeff/monitor_test3,m) exten => 5043, n,
changemonitor(monitor_test4) exten => 5043, n, dial(sip/5001)
exten => 5044, 1, monitor(wav,monitor_test4,m) exten => 5044, n,
changemonitor(tmp/jeff/monitor_test5) ; this looks to fail by
design and emits a warning exten => 5044, n, dial(sip/5001)
........ ................
2010-02-25 22:42 +0000 [r248949] Mark Michelson <mmichelson@digium.com>
* /, main/acl.c: Merged revisions 248946 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r248946 |
mmichelson | 2010-02-25 16:41:48 -0600 (Thu, 25 Feb 2010) | 5
lines Fix incorrect ACL behavior when CIDR notation of "/0" is
used. AST-2010-003 ........
2010-02-25 21:25 +0000 [r248864] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c, /: Merged revisions 248861 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r248861 | tilghman | 2010-02-25 15:22:39 -0600 (Thu, 25 Feb 2010)
| 22 lines Merged revisions 248859 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r248859 | tilghman | 2010-02-25 15:21:05 -0600 (Thu, 25 Feb 2010)
| 15 lines Some platforms clear /var/run at boot, which makes
connecting a remote console... difficult. Previously, we only
created the default /var/run/asterisk directory at install time.
While we could create it in the init script, that would not work
for those who start asterisk manually from the command line. So
the safest thing to do is to create it as part of the Asterisk
boot process. This also changes the ownership of the directory,
because the pid and ctl files are created after we setuid/setgid.
(closes issue #16802) Reported by: Brian Patches:
20100224__issue16802.diff.txt uploaded by tilghman (license 14)
Tested by: tzafrir ........ ................
2010-02-25 18:52 +0000 [r248797] Jeff Peeler <jpeeler@digium.com>
* res/res_monitor.c, /: Merged revisions 248793 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r248793 | jpeeler | 2010-02-25 12:37:56 -0600 (Thu, 25 Feb 2010)
| 22 lines Merged revisions 248757 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r248757 | jpeeler | 2010-02-25 12:06:54 -0600 (Thu, 25 Feb 2010)
| 15 lines Ensure that monitor recordings are written to the
correct location. Recordings should be placed in the monitor
directory when a non-absolute path is used. Exact dialplan used
for testing: exten => 5040, 1,
monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n,
dial(sip/5001) exten => 5041, 1,
monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n,
dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b)
exten => 5042, n, dial(sip/5001) ABE-2101 ........
................
2010-02-24 21:29 +0000 [r248643] Tilghman Lesher <tlesher@digium.com>
* /, main/logger.c: Merged revisions 248584 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r248584 | tilghman | 2010-02-24 15:17:26 -0600 (Wed, 24 Feb 2010)
| 14 lines Merged revisions 248582 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r248582 | tilghman | 2010-02-24 15:02:18 -0600 (Wed, 24 Feb 2010)
| 7 lines Remove color code sequences from verbose messages that
go to logfiles. (closes issue #16786) Reported by: dodo Patches:
logger2.patch uploaded by dodo (license 989) Tested by: tilghman
........ ................
2010-02-23 16:37 +0000 [r248398] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 248397 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r248397 | dvossel | 2010-02-23 10:34:39 -0600 (Tue, 23 Feb 2010)
| 15 lines Merged revisions 248396 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r248396 | dvossel | 2010-02-23 10:26:05 -0600 (Tue, 23 Feb 2010)
| 9 lines fixes invite with replaces deadlock (closes issue
#16862) Reported by: pwalker Patches: replaces_deadlock_1.4
uploaded by dvossel (license 671) Tested by: pwalker, dvossel
........ ................
2010-02-19 19:07 +0000 [r248011] Tilghman Lesher <tlesher@digium.com>
* channels/chan_console.c, main/loader.c, /: Merged revisions
228798 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r228798 |
tilghman | 2009-11-09 01:37:52 -0600 (Mon, 09 Nov 2009) | 14
lines Fix various problems detected with Valgrind. * chan_console
accessed pvts after deallocation. * The module loader did not
check usecount on shutdown, which led to chan_iax2 reading a
timer that was already unloaded. (closes issue #16062) Reported
by: alexanderheinz Patches: 20091109__issue16062.diff.txt
uploaded by tilghman (license 14) Tested by: tilghman ........
2010-02-19 19:00 +0000 [r248005] Moises Silva <moises.silva@gmail.com>
* channels/chan_dahdi.c, /: Merged revisions 248003 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r248003 | moy | 2010-02-19 13:38:34 -0500 (Fri, 19 Feb 2010) | 1
line mfcr2 issue 0016844 - Fix portability bit fields and make
mfcr2_immediate_accept work again, reported and patched by
korihor ........
2010-02-19 18:45 +0000 [r248004] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c, /: Merged revisions 247914 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r247914 | rmudgett | 2010-02-19 11:33:33 -0600
(Fri, 19 Feb 2010) | 62 lines Merged revisions 247910 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
...........