David Vossel [Mon, 10 May 2010 18:36:10 +0000 (18:36 +0000)]
fixes crash in chan_console
There is a race condition between console_hangup()
and start_stream(). It is possible for console_hangup()
to be called and then the stream thread to begin after the hangup.
To avoid this a check in start_stream() to make sure the pvt-owner
still exists while the pvt lock is held is made. If the owner
is gone that means the channel hung up and start_stream should
be aborted.
Alec L Davis [Fri, 7 May 2010 23:54:15 +0000 (23:54 +0000)]
VoicemailMain and VMauthenticate, allow escape to the 'a' extension when a single '*' is entered
Where a site uses VoicemailMain(mailbox) the users have to be at their own extension to clear
their voicemail, they have no way of escaping VoicemailMain to allow entry of new boxnumber.
This patch, allows a site to include to 'a' priority in the VoicemailMain context, to allow an escape.
If the 'a' priority doesn't exist in the context that VoicemailMain was called from then it acts as the old behaviour.
Tilghman Lesher [Fri, 7 May 2010 20:35:17 +0000 (20:35 +0000)]
Use the detected pthread building flags in every place, instead of hardcoding -lpthread.
We nicely detect the right flags on each system for building Asterisk with
pthreads, then ignore it for every other build option that requires us to
build with pthreads. This caused some items to return a false negative.
Also cleanup some minor naming issues that caused "library library" redundancy
in the output.
Jeff Peeler [Fri, 7 May 2010 15:33:52 +0000 (15:33 +0000)]
Fix deadlock in sig_pri when hanging up.
The pri_dchannel thread currently violates locking order by locking the private
and then attempting to queue a frame, which needs to lock the channel. Queueing
a frame is unneccesary though and is actually a regression since sig_pri.
All the places that currently use ast_softhangup_nolock now will just set the
softhangup value directly as before.
Russell Bryant [Thu, 6 May 2010 13:58:07 +0000 (13:58 +0000)]
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.
David Vossel [Wed, 5 May 2010 19:13:57 +0000 (19:13 +0000)]
fixes sip native transfer
The Refer-To header field containing the Replaces header in the URI
was not being decoded properly. This caused invalid parsing between
the caller id field and the domain resulting in a failed transfer.
Mark Michelson [Wed, 5 May 2010 18:28:05 +0000 (18:28 +0000)]
Prevent unnecessary warnings when getting rtpsource or rtpdest.
If a recognized media type was present, but the media type was not
enabled for the channel, then a warning would be emitted. For instance,
attempting to get CHANNEL(rtpsource,video) on a call with no video would
cause a warning message to appear.
With this change, the warning will only appear if the stream argument
is not recognized as being a media type that can be specified.
Paul Belanger [Wed, 5 May 2010 00:22:32 +0000 (00:22 +0000)]
New static asterisk.conf.sample file.
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.
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.
Jeff Peeler [Mon, 3 May 2010 22:13:24 +0000 (22:13 +0000)]
Add new admin features to meetme: Roll call, eject all, mute all, record in-conf
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
Avoid making AstData depend on libxml2 to compile.
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.
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.
Richard Mudgett [Fri, 30 Apr 2010 22:24:28 +0000 (22:24 +0000)]
Separate the uses of NUM_DCHANS and MAX_CHANNELS into PRI, SS7, and MFCR2 users.
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[].
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.
Mark Michelson [Fri, 30 Apr 2010 19:53:36 +0000 (19:53 +0000)]
Fix logic reversal error when queue callers join the queue.
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.
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. Also introduce an additional
optimization, in the form of using the CLOEXEC flag to close descriptors at
the right time.
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
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.
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.
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.
........
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).
........
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.
* 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.
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
........
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.
Update res_fax and res_fax_spandsp to be compatible with Fax For Asterisk 1.2.
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.
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)
........
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().
Russell Bryant [Thu, 22 Apr 2010 21:06:53 +0000 (21:06 +0000)]
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.
Jason Parker [Thu, 22 Apr 2010 19:08:01 +0000 (19:08 +0000)]
Remove ABI differences that occured when compiling with DEBUG_THREADS.
"Bad Things" would happen if Asterisk was compiled with DEBUG_THREADS, but a
loaded module was not (or vice versa). This also immensely simplifies the
lock code, since there are no longer 2 separate versions of them.
This module implements an abstraction for retrieving and exporting
asterisk data.
Developed by:
Brett Bryant <brettbryant@gmail.com>
Eliel C. Sardanons (LU1ALY) <eliels@gmail.com>
For the Google Summer of code 2009 Project.
Documentation can be found in doxygen format and inside the
header include/asterisk/data.h
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).
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.
David Vossel [Wed, 21 Apr 2010 18:13:36 +0000 (18:13 +0000)]
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.
Added a new manager command to mute/unmute MixMonitor audio on a channel.
Added a new feature to audiohooks so that you can mute either read / write
(or both) types of frames - this allows for MixMonitor to mute either side
of the conversation without affecting the conversation itself.
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.
Updated the doxygen \arg line after looking at the file for some other Asterisk documentation
and noticing they weren't up to date. Thanks to seanbright for looking at the code for me :)
projects / thirdparty / asterisk.git / log
