This simply moves the functionality from the Makefile (cleaning it up) into an external
asterisk.conf.samples file. Also updates formatting (easier to read) and grammar
changes to asterisk.conf.samples.
(closes issue #17027)
Reported by: pabelanger
Patches:
0017027.asterisk.conf.v6.patch uploaded by pabelanger (license 224)
Tested by: qwell, lmadsen, pabelanger, chappell
Review: https://reviewboard.asterisk.org/r/616/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261095 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The CLI "dahdi show channel" command was not correctly reporting the
InAlarm status.
The inalarm flag is now consistently passed between chan_dahdi and
submodules.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260924 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the following in-conference admin DTMF features:
*81 - Roll call (or simply user count if INTROUSER isn't enabled)
*82 - Eject all non-admins
*83 - Mute/unmute all non-admins
*84 - Start recording the conference on the fly
FWIW, this code uses newly recorded prompts.
(closes issue #16379)
Reported by: rfinnie
Patches:
meetme-enhancements-232771-v1.patch uploaded by rfinnie (license 940)
modified slightly by me
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
We have some functions inside the AstData API to get the tree
in XML form, but it is not required at the moment to compile
asterisk and we can disable that part of the API if we don't have
libxml2 support.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260521 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260437 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Created
SIG_PRI_MAX_CHANNELS, SIG_PRI_NUM_DCHANS
SIG_SS7_MAX_CHANNELS, SIG_SS7_NUM_DCHANS
SIG_MFCR2_MAX_CHANNELS
Also fixed the declaration of pollers[] in mfcr2_monitor(). It was
dimensioned to the number of bytes in struct dahdi_mfcr2.pvts[] and not to
the same dimension of the struct dahdi_mfcr2.pvts[].
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a specific position is specified for the queue, the idea
was that the caller cannot be placed ahead of higher-priority
callers. Unfortunately, the logic was reversed so that the caller
could ONLY be placed ahead of higher priority callers.
Discovered while writing a unit test.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This saves time, when, even though the system allows the process limit to be
that high, the practical limit is much lower. Also introduce an additional
optimization, in the form of using the CLOEXEC flag to close descriptors at
the right time.
(closes issue #17223)
Reported by: dbackeberg
Patches:
20100423__issue17223.diff.txt uploaded by tilghman (license 14)
Tested by: dbackeberg
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260292 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260231 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259870 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259853 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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).
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259538 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259439 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259353 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259307 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259023 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(closes issue #16988)
Reported by: frawd
Patches:
chan_sip_sdp_verbose_fix.diff uploaded by frawd (license 610)
Tested by: russell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258934 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The fax session initilization code for T.38 faxes has been rewritten. T.38 session initialization was removed from generic_fax_exec, and split into two different code paths for receive and send. Also the 'z' option (to send a T.38 reinvite if we do not receive one) was added to sendfax.
In the output of 'fax show sessions', the 'Type' column has been renamed to 'Tech' and replaced with a new 'Tech' column that will report 'G.711' or 'T.38'.
Control of ECM defaults has been added to res_fax
A 'fax show settings' CLI command has been added.
Support of the new AST_T38_REQUEST_PARMS control method request to handle channels that have already received a T.38 reinvite before the FAX application is start has been added.
Support for the 'fax show settings' command has been added to res_fax_spandsp and handling of the ECM flag has been slightly altered.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258896 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258776 65c4cc65-6c06-0410-ace0-fbb531ad65f3
