https://origsvn.digium.com/svn/asterisk/branches/1.2
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r69469 | qwell | 2007-06-14 18:21:45 -0500 (Thu, 14 Jun 2007) | 4 lines
Fix an issue where the line number in an unterminated comment block error message would show the wrong line number.
"Reported" to me on #asterisk (somebody posted an error message, and I happened to catch it)
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formats already match up. There are code paths that call this function on a
pair of channels multiple times. This made calls fail that were using g729
in some cases. The reason is that codec_g729a will unregister itself from the
list of available translators will all licenses are in use. So, the first
time the function got called, the right translation path was allocated.
However, the second time it got called, the code would not find a translation
path to/from g729 and make the call fail, even if the channel actually already
had a g729 translation path allocated.
(SPD-32)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r67715 | russell | 2007-06-06 11:40:51 -0500 (Wed, 06 Jun 2007) | 5 lines
We have some bug reports showing crashes due to a double free of a channel.
Add a sanity check to ast_channel_free() to make sure we don't go on trying
to free a channel that wasn't found in the channel list.
(issue #8850, and others...)
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bogus on my machine. ast_safe_string_alloc() was broken. It called
vsnprintf() on a va_args list twice without re-initializing it. After the first
usage, va_end() and va_start() must be called again.
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all of the modules. "stop now" is considered a non-graceful shutdown and will
not go through this process.
(issue #9804, reported by chrisost, patch by me)
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oscillating and incorrect data. Additionally, the RTT would sometimes report
negative values due to incorrect calculations.
(issue #9601, patch from davetroy)
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I realize that there are other ways to get this,
but we really don't need to just show it in plain text so easily.
Issue 9273, patch by junky
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either because the Challenge action was never issued, or some other reason,
give a proper error message and return an error instead of claiming that the
user wasn't found.
(reported by jsmith on IRC)
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code in that if a channel does not have a send_digit_begin() callback, it only
cares about DTMF END events. (pointed out by Michael Neuhauser on the
asterisk-dev list)
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send_digit_begin() callback. Checking the END_DTMF_ONLY flag was the
wrong thing to do, because that flag indicates that a *bridged* channel
only wants DTMF END events coming from this channel.
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This set of changes came from a debugging session I had with Dwayne Hubbard.
When he called into his home FXO, ran the Echo application, and pressed a
digit, the digit would be echoed back and would never end. This is fixed,
along with a couple other little improvements.
* When chan_zap is in the middle of playing a digit to a channel, it feeds
back null frames, not voice frames. So, I have modified ast_read to check
the timing on emulated DTMF when it receives null frames, in addition to
where it was doing this on voice frames.
* Make a tweak to setting the duration on emulated DTMF digits. If there was
no duration specified, it set it to be the minimum, instead of the default.
* Instead of timing the emulated digits off of the number of samples in audio
frames that pass through, just use time values. Now there is no code in this
section that assumes 8kHz audio.
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Fix some issues related to generating inband DTMF. There are two changes here:
1) The list of DTMF tones in the senddigit_begin() function explicitly
specified 100ms of the tone followed by 100ms of silence. This really
broke things with the way that Asterisk now wants complete control
over when the digit begins and ends. So, regardless of what Asterisk
really wanted to do, this was going to play out the tone at the length it
wanted to. This caused various problems like DTMF translation to inband to
be extremely unreliable.
The list of tones has been changed so that the correct DTMF tone is played
indefinitely until Asterisk tells it to stop.
2) ast_write() had to be modified to let a DTMF_END frame get processed even
when a generator is present. This is how the tone will finally get stopped.
(issues #8944, #9250, #9348, maybe others. Thanks to mdu113 from #8944 for
the testing and feedback!)
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don't even bother creating a temporary bogus channel, since that is only for
allowing certain functions to operate on the variables as if they were on a
channel. Most importantly, this fixes a crash.
(issue #9613, reported by callguy, fixed by me)
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the asterisk-dev mailing list. I changed the enforced minimum length of a
digit from 100ms to 80ms. Furthermore, I made it now enforce a gap of 45ms in
between digits. These values are not configurable in a configuration file
right now, but they can be easily changed near the top of main/channel.c.
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will get notified of these changes even when an owner channel is not provided.
This isn't from a specific bug report, it's just something I noticed while
poking around.
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channel. So, this little hack lets them work in places where a channel doesn't
exist, such as within DUNDi configuration.
(issue #9465, reported and patched by Corydon76, testing by blitzrage)
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Per comment from Dave Troy:
This adds back in some simple typecasting I had in an earlier version
which I realize now may be breaking things.
Issue #9554.
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However, after much discussion, it has been decided that adding this to 1.4 is
not in the best interests of the project. It has been removed here, but will
remain in trunk.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r60849 | tilghman | 2007-04-08 21:49:06 -0500 (Sun, 08 Apr 2007) | 2 lines
Don't check for error when lowering priority (according to the manpage, it should never happen anyway). It might could happen, though, if another thread messed with the priority, so safeguard against that (reported via -dev list).
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Asterisk GUI project, we need a fully functional HTTP interface with access
to the Asterisk manager interface. One of the things that was intended to be
a part of this system, but was never actually implemented, was the ability for
the GUI to be able to upload files to Asterisk. So, this commit adds this in
the most minimally invasive way that we could come up with.
A lot of work on minimime was done by Steve Murphy. He fixed a lot of bugs in
the parser, and updated it to be thread-safe. The ability to check
permissions of active manager sessions was added by Dwayne Hubbard. Then,
hacking this all together and do doing the modifications necessary to the HTTP
interface was done by me.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r60134 | russell | 2007-04-04 12:38:47 -0500 (Wed, 04 Apr 2007) | 6 lines
It is valid to redirect channels via the manager interface that are not in the
UP state. Instead of checking for that to prevent to ensure a dead channel
doesn't get redirected, just use the ast_check_hangup() API call.
(issue #9457, reported by Callmewind, patch by me)
(related to issue #8977)
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r59886 | russell | 2007-04-03 12:58:19 -0500 (Tue, 03 Apr 2007) | 5 lines
When doing a built-in blind or attended transfer, restore the ability to use '#'
to terminate the number and immediately do the transfer instead of having to
dial the number and just wait for the feature digit timeout.
(issue #8366, xueliangliang)
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r59608 | russell | 2007-04-01 17:35:25 -0500 (Sun, 01 Apr 2007) | 6 lines
Add the SO_REUSEADDR flag to sockets handled by netsock. This is needed by
the patch that went in for issue 7874. chan_iax2 needs to be able to create
socket that is lisetning on INADDR_ANY, but also be able to bind sockets to
specific addresses. (Thanks to Stevenson on the asterisk-dev mailing list
for explaining why this flag was needed.)
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r59357 | russell | 2007-03-29 12:14:33 -0500 (Thu, 29 Mar 2007) | 5 lines
If an error occurs when reading from an RTP socket, and the error code does not
indicate that we should try again, then return NULL instead of a "null frame".
This will prevent Asterisk from trying over and over again, and eventually
causing the system to crash. (issue #8285, john)
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because they get set in sip_hangup. So, there are common situations where
the variables will not be available in the dialplan at all. So, this patch
provides an alternate method for getting to this information by introducing
AUDIORTPQOS and VIDEORTPQOS dialplan functions.
(issue #9370, patch by Corydon76, with some testing by blitzrage)
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* Convert most of the doc directory into a single LaTeX formatted document
so that we can generate a PDF, HTML, or other formats from this
information.
* Add a CLI command to dump the application documentation into LaTeX format
which will only be include if the configure script is run with
--enable-dev-mode.
* The PDF turned out to be close to 1 MB, so it is not included. However, you
can simply run "make asterisk.pdf" to generate it yourself. We may include
it in release tarballs or have automatically generated ones on the web site,
but that has yet to be decided.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r56504 | russell | 2007-02-23 17:20:55 -0600 (Fri, 23 Feb 2007) | 8 lines
Fix up a couple more signal handlers to not do bad things that could cause
various undesirable results. The other day, I made Asterisk deadlock by
hitting Control-C because of a bad signal handler. Now, signal handlers
just set a flag and write to an alert pipe for the flag to be handled. Then,
there is another thread that is monitoring for these flags. If being run in
console mode, it is just the main thread. If Asterisk is in the background,
a thread is created to do it.
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This batch of changes to the SLA code does a few different things.
* I made the SLA code event driven instead of having to act in a lot of busy
loops while dialing things to wait for state changes. This makes the code
more efficient and readable at the same time.
* I have implemented a couple of new features. The first is inbound trunk
ringing timeouts. This is an option that defines how long to let an incoming
call on a trunk to ring.
* I have also implemented ring timeouts for stations. They may be specified
for the entire station, meaning it is how long to let the station ring before
giving up. You can also specify a ring timeout for a specific trunk on a
station. So, you can say that you only want a specific station to ring 5
seconds if it is line1 ringing, but otherwise, there is no timeout.
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This is a completely new implementation of the SLA functionality introduced in
Asterisk 1.4. It is now functional and ready for testing. However, I will be
adding some additional features over the next week, as well.
For information on how to set this up, see configs/sla.conf.sample
and doc/sla.txt.
In addition to the changes in app_meetme.c for the SLA implementation itself,
this merge brings in various other changes:
chan_sip:
- Add the ability to indicate HOLD state in NOTIFY messages.
- Queue HOLD and UNHOLD control frames even if the channel is not bridged to
another channel.
linkedlists.h:
- Add support for rwlock based linked lists.
dial.c:
- Add the ability to run ast_dial_start() without a reference channel to
inherit information from.
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ASTCFLAGS, instead of at the end. This way, we ensure that we find the local
headers first before accidentally trying to use headers that exist in
locations specified in the ASTCFLAGS passed from the main Makefile.
(issue #8637, ovi)
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because we need to make sure that its configure script gets executed again,
because the CFLAGS we want to pass to editline may have changed.
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take the low 16 bits of the integer part, and the high 16 bits of the
fractional part. However, the code here was erroneously taking the low 16 bits
of the fractional part. It then shifted the result 16 bits down, so the result
was always zero. This fix makes it grab the appropriate high 16 bits, instead.
(issue #8991, pointed out by andre_abrantes)
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Jason Parker
Kevin P. Fleming
Russell Bryant
Joshua Colp
Steve Murphy
Tilghman Lesher
Olle Johansson
Doug Bailey
Dwayne M. Hubbard
Paul Cadach