SVN commits to the Zaptel project wrote:
* Start documenting module parameters in the README.
There is already a documentation file for module parameters, in
doc/module-parameters.txt.
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, with operations that are meant to decrement the refcount.
Umm... s/expanation/explanation/ :-)
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that ztscan is in the Zaptel tree it's easy to upgrade them together.
On the subject of supporting DACS; any digital span with more than one
channel should support DACS mode if at all possible. It's the only way
to do zero- (or minimal) latency digital switching between channels.
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;
__zt_getbuf_chunk(chan, buf);
Shouldn't these be unlikely() instead of likely()? Even if someone
enables this code, it's still far more likely that their channels will
be in 'I need data' mode than not, especially since none of the
low-level drivers currently support this flag.
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response without SDP means Asterisk should not send audio
*to* the phone, but it does not impact Asterisk receiving audio.
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Tzafrir Cohen wrote:
It would not help a bit if all of your channels go thrrough digital
Zaptel channels.
OK, I see your point. It's a micro-optimization anyway.
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like to use.
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logic you choose and then let
app_queue monitor those to determine the availability of queue members.
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want to play around with the
drivers on my system and ensure they behave identically before and after
the change... if so, I'll remove the generated tones and the background
thread(s) that feed them out.
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Adrià Vidal wrote:
have $ svn co http://svn.digium.com/svn/asterisk/team/russell/chan_console
dead?
trying to test it into my macbook too...
It's been merged into SVN trunk already, you can just test the trunk
instead.
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we should :-)
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spec is
obviously based on a phone on an analogue line generating the DTMF. Have
you been testing there with analogue or digital connections?
The customer who requested the ability to do this has been doing
regulatory compliance testing with analog circuits in Brazil.
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instead. This will make it
very easy for an admin to change the location for one item.
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echo
INPUT (../$${file}) $@; done
@for file in $(patsubst %,$(SUBDIR)/%,$(filter-out %.eo,$^)); do echo
INPUT (../$${file}) $@; done
That will probably work, but it'll make the linker script a lot larger
if the path is long :-)
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Tzafrir Cohen wrote:
This is generic zaptel code, rather than zaptel-specific.
Implement
zt_span_from_lineconfig(struct zt_span *span, struct lineconfig *lc)
?
I don't understand the question... can you rephrase this? What would
this function do?
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1.6's lifetime, rather than having to wait for 1.8.
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that is 'consuming' the frames. If ast_write() was the
consumer, then you'd never be able to write the frame to more than one
location without duplicating it, which would be needless overhead.
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chan_sip has fed you a frame of audio data,
why should it care who frees it, and more importantly, how could it
possibly know that the frame is no longer needed?
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guidelines mandate.
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Johansson Olle E wrote:
Join me in congratulations to Philippe for this election!
Fantastic! Glad to hear it Philippe!
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.eo and .eoo are handled exactly in the same way.
I am under the impression that Makefile.rules should have this change
-%.eoo: %.o
+%.eo: %.oo
I believe this is correct, and should result in the proper linking of
the modules.
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constructors are automatically executed when
the main 'asterisk' binary is executed, and they get registered on the
loader's list of available modules, but their load_module() callbacks
are not executed until a 'load' is requested for that module.
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was unclear. I was agreeing with your proposed simplification;
there is no reason for .oo and .o files to be handled differently when
being embedded.
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SVN commits to the Asterisk project wrote:
On linux the order of execution of constructor
was evidently different (it may depend on the
ordering of modules in the ELF file).
On Linux, mutexes do not require constructors for initialization, so
this has never been an issue.
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make the changes directly there
if you feel they are worth making.
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.
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' declaration of it in this header file. This would allow
these two functions to then be static and local to that module.
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changes more frequently than
once per major release, so it is important that as part of this change
we do a very good job both justifying and documenting API changes
between minor releases.
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the new features or other
more invasive changes that are present in the newer releases.
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will still be populated by contractor vehicles, moving vans
and others who don't have anything to do with Digium directly...
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with a copy of -interface if no member name is provided
when the member is added to the queue.
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and didn't see that happening
in create_member()... but it would be a good way to solve this problem
(and eliminate a bunch of conditionals that already exist).
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it be done at the lowest level
(create_member) instead of in the higher places that end up calling
create_member, but what we have now works.
Given that, I've removed the conditional logic that I put in yesterday
since it isn't necessary.
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we shouldn't do the same
thing in chan_iax2.
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a 'feature submission', since you are offering an implementation.
What we don't want on the issue tracker is a an issue being opened
saying 'Asterisk should have XYZ' and that's all :-)
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and switched to macros in all cases; there
is no need for the inline function version, as any debugging code that
is needed can be put into the macros using the 'do { } while(0)' trick.
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... as it turns out, this is already fixed :-)
You see, part of this change is that the file listings and downloads are
actually being served up by a PHP script, not Apache in all its raw
glory... and the script does not truncate the names.
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.
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on this behavior and would break badly
if the mutexes were not recursive. This is documented in a comment very
early in include/asterisk/lock.h, so I'm not sure why you came to the
conclusion that non-recursive mutexes were in use.
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sethdlc-new to sethdlc.
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see any reason to install it by default.
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by zaptel devices too
regular or unsuitable for this purpose ?
They are *completely* regular, 1000 times per second per card. Not a
good entropy source at all :-)
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the odd key.
I think the issue here is that in IAX2, the *entire* packet is
encrypted, there is no unencrypted header that can indicate which key
was used for encryption.
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and problems caused by the current Realtime implementation.
Hopefully in the next few days someone will have time to document what
we talked about at the DevCon and keep the discussion going on this list :-)
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Vadim Lebedev wrote:
The trick is that the pipe is ALWAYS empty And when we push
something to the pipe, we pull the data from it immideately afterwards.
So you've replaced a pair of read()/write() calls with two calls to
splice()? It's the same number of userspace/kernelspace boundary
Vadim Lebedev wrote:
Now, when we agree that it IS possible to use single pipe for multiple
bridges
let's look on benefits:
I did not agree that it was possible, I told you it seemed possible but
would likely provide little or no benefit.
As for now Asterisk when packet arrives on a briged
Vadim Lebedev wrote:
You're right but only in the case that when you have separtae briding
threads for each direction.
I was thinkin about situation when there is ONE briging theard for two
directions. The you can you the same pipe
for both direction.
No, you cannot. How are you going to
[EMAIL PROTECTED] wrote:
Modified: branches/1.4/formats/format_wav.c
URL:
http://svn.digium.com/view/asterisk/branches/1.4/formats/format_wav.c?view=diffrev=60325r1=60324r2=60325
==
---
Olle E Johansson wrote:
But what's the format of the attachment? Back to codec negotiation
again. I would call it call properties negotiation
because it is more than codecs - it will soon also become security
properties.
Given the importance and complexity of this issue, I am already making
Yuan Qin wrote:
The ast_log() will lock a mutex if appropriate, but the mutex may be in
locked state already.
Maybe we should use pthread_atfork() instead of fork() or never call
some functions that hold mutex
before execv() in child process.
pthread_atfork() does not fork, it does
Tim Panton wrote:
If I understand the issue right, we need to be able to re-negotiate the
parameters mid-call (change the quality measure on a codec for example)
not just at setup time.
That is only part of the issue; the more immediate problem is that we an
Asterisk 'format' encompasses the
Tzafrir Cohen wrote:
On Fri, Mar 23, 2007 at 01:43:26AM -, svn-commits@lists.digium.com wrote:
Author: kpfleming
Date: Thu Mar 22 20:43:26 2007
New Revision: 2334
URL: http://svn.digium.com/view/zaptel?view=revrev=2334
Log:
remove pointless file ID lines
Why are the file IDs
Clod Patry wrote:
Is Digium willing to pay the 'extra-sounds' in french with June Wallack?
I doubt it. We haven't even had them re-recorded by Allison in
high-quality format.
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Admin DeryTelecom wrote:
I am talking about a SIP Dial and not a Zap Dial, it seems that this
option is available on Zap Dial.
Then this will be the responsibility of the SIP endpoint, not Asterisk.
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Anthony Lamantia wrote:
obvious reasons .. ?, I really would like to know what the risk to my
asterisk servers are.
We have never, and will never, help potential exploiters directly.
The issue is that a very simple SIP packet can cause Asterisk to crash.
Figuring out how to construct that
Matthew Rubenstein wrote:
This security reality is well known in the programming industry. I'm
disappointed to see Digium acting as if it weren't.
What is obscured? We clearly stated that the vulnerability existed, the
patch to fix it was public, the release that contained that patch was
Daniel Gonzalez wrote:
What is the position of the Asteriks people regarding this issue? In
case you have interest, we welcome suggestions on how to integrate the
Zaptel library. Currently we are thinking about using the
drivers/telephony subdirectory for it.
Please review the message
Center), and will be low-key and open only to
serious developers and contributors. We are expecting to keep the
attendance to 50 people or less, including the entire Digium Asterisk
development team (currently around 10 people).
If you wish to participate, please contact Kevin P. Fleming so he can
Patrick wrote:
With asterisk 1.4svn from yesterday I have tested --with-imap on a FC6
and CentOS4 box. Currently it fails although the required files are
installed (libc-client-devel package 2004g version):
/usr/lib/c-client.a
/usr/lib/libc-client.a
/usr/lib/libc-client.so
Patrick wrote:
checking linux/soundcard.h usability... yes
checking linux/soundcard.h presence... yes
checking for linux/soundcard.h... yes
Should configure not skip this?
It should.
checking h323.h usability... no
checking h323.h presence... no
checking for h323.h... no
Should
Olle E Johansson wrote:
Please test this. If it works, we might be able to consider this a bug
fix and include it
in 1.4.
+1 from me
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Tomislav Parèina wrote:
Is uniqueid globally unique? I have three Asterisk installations and I need
to store data from all of them in same database, in same table. Will this
uniqueid field be unique?
Please search the list archives before posting questions. This topic was
just talked about
Paul Cadach wrote:
As I can see there is missed synchronization between origsvn and
svn.digium.com for last 2 days. Could someone take a
look please?
This has finally been corrected; the public SVN mirror is now up to date
and will stay that way unless I break it again :-(
Oded Arbel wrote:
The problem I'm having, is that I don't think that a channel's uniqueid
is unique across multiple Asterisk installations - under some very
common behaviors of a multiple Asterisk installation, some channels
created on different Asterisk instances will have the same uniqueid.
Ron Joffe wrote:
Are we heading down the wrong path with the flash command ?
Yes. There is no such thing as a 'hook-flash' on PRI, since hook-flash
is part of analog signaling.
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John Todd wrote:
While I understand the sentiment here, I'm not sure this is a good
idea. This builds in a 500ms post-dial delay issue into every call.
I've been building systems for three years now, and everywhere there is
an Answer (which, I believe, should be the only method that picks up
Jared Smith wrote:
Wouldn't this be better served as an argument to the Answer()
application? We already have one argument for a delay *before*
answering the channel, so why not have one for a delay *after*
answering the channel. Now the commit log says when a channel gets
automatically
[EMAIL PROTECTED] wrote:
Author: file
Date: Fri Jan 12 21:26:04 2007
New Revision: 50676
URL: http://svn.digium.com/view/asterisk?view=revrev=50676
Log:
Merge in trunkrealloc option for chan_iax2. (issue #8267 reported by
marcodmb, branch by anthonyl)
Wouldn't it be clearer (and just as
Tony Mountifield wrote:
If the module can't automatically distinguish between revisions of card,
then perhaps it should be told, using an option parameter, so that the
same code can be used with both older and newer revisions of the card?
That card has not been shipped to anyone except beta
[EMAIL PROTECTED] wrote:
Author: oej
Date: Tue Jan 2 07:50:51 2007
New Revision: 49152
URL: http://svn.digium.com/view/asterisk?view=revrev=49152
Log:
Update sample config
Modified:
trunk/ (props changed)
trunk/configs/features.conf.sample
Propchange: trunk/
Brian Capouch wrote:
In other words, how does one know that pg_config ought to be built for
the host instead of the target--I'm not super-familiar with it. Perhaps
its only function in life is to inform build scripts?
Yep, that's exactly it. If the binary being built has its only purpose
as a
Brian Capouch wrote:
In other words, how does one know that pg_config ought to be built for
the host instead of the target--I'm not super-familiar with it. Perhaps
its only function in life is to inform build scripts?
Side note: PostgreSQL is one of the few packages that doesn't just a
Florian Overkamp wrote:
Is there anything against just enabling this in additional headers when
cdr_manager is enabled ? Seems closely related to that function...
It's not at all related to CDRs, so that doesn't seem to make any sense.
We also try to avoid compile-time options that add/remove
Brett Crapser wrote:
Now the above are only available in the 2.6 kernel but there are no
if statements around them so are you dropping 2.4 kernel support or
is this just a oops?
This is trunk; it's a development area, which means very often it will
be an inconsistent state.
ztcodec_dte is an
Luigi Rizzo wrote:
I'd suggest to revert this and, if necessary, make the
change to codecs/codec_zap.c
I'm not even sure why this is needed, when we can already test for the
proper things in the configure script.
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Nic Bellamy wrote:
In the books I have on POSIX threads, plus the manpages for various
(g)libc functions, re-entrant is the term used to describe functions
that can be called concurrently from multiple threads without problems
(and conversely, things like ctime() - this function is not
Russell Bryant wrote:
A reentrant function:
- Does not hold static data over successive calls
OK, given this definition I'd agree, but I'm concerned that the more
common understanding of non-reentrant functions (in most people's minds)
is that the function is also not thread-safe, but in
Tzafrir Cohen wrote:
It's useful for people who already have an account on the Digium SVN. Not
much of help otherwise. And as it is not the same repository, it is not
even simple to merge bugs and use 'svn diff'.
Anyone who wants to host projects there can ask for an account and do
so; there
Johansson Olle E wrote:
Kevin has been working on installing new servers today, so I guess that the
problem comes from that. Hopefully it will mean that we have
extra capacity soon :-)
Actually, that was unrelated... the new servers are at a different facility.
F. Mitchell Felling II wrote:
I have a need to not only launch, but control app_dictate from the
Asterisk Management Interface. I expected this to be rather simple by
using the playdtmf action. However, it seems the DTMF is only played to
the channel (extension) not to the instance of the
Russell Bryant wrote:
You're probably right. I'm starting to feel that the best thing to do
at this
point is to just make the way Asterisk does the matching not be completely
insane. The performance benefit is probably not worth trying to deal with
making such a significant change in the
Di-Shi Sun wrote:
1. What is the IE identification number we should propose? IAX2 has used
0x01 to 0x33. Corydon76 had posted a patch on bugs.digium.com that used
0x32. We wonder if there is any suggestion for how to use this
identification number resource.
We don't have any formal process
[EMAIL PROTECTED] wrote:
Author: rizzo
Date: Wed Nov 15 08:11:28 2006
New Revision: 47652
URL: http://svn.digium.com/view/asterisk?view=revrev=47652
Log:
update the internal cli api following comments from kevin.
This change basically simplifies the interface of the
new-style handler
Luigi Rizzo wrote:
I am afraid the e = argv[-1] trick is probably something that we need
to keep for a while to help old-style handlers, but that is a relatively
straightforward, and hopefully sufficiently documented to avoid confusion.
Why do we need this for old style handlers? They never
Corrado Santoro wrote:
I'm writing an application that, a part from the main thread (say A),
has another thread (say B). I have to perform a communication of
events(+data) from thread B to thread A, and I'm going to do this with a
classical synchronized circular queue. Thus I'm wondering if
Luigi Rizzo wrote:
the question was aimed to know if whoever wrote/is familiar
with that part of the code knows why - whether this is a residue
of some old code (likely), or there is something that got
deleted elsewhere in the code justifying the deletion.
Well, finding the person who wrote
[EMAIL PROTECTED] wrote:
Author: file
Date: Tue Nov 7 23:08:31 2006
New Revision: 1570
URL: http://svn.digium.com/view/zaptel?rev=1570view=rev
Log:
Don't build the firmware headers unless needed. This shaves ~3.5 seconds off
build time.
Actually, we need to do this all the way back to
Tzafrir Cohen wrote:
Note that I don't aim that high. I just aim at keeping this option open
(let alone making 'reload chan_zap.so' work as defined).
As the US Air Force recruitment ads say: AIM HIGH!
Seriously... lots of people will be very happy if someone out there can
find the time to
Luigi Rizzo wrote:
well, the thing is, on FreeBSD at least libnsl does not exist so
putting it in the loader flags breaks the build.
The same may happen for other platforms.
i would suggest to remove these extra libs (except for gnutls, but probably it
is already there after detecting gnutls)
Russell Bryant wrote:
That macro is provided when you install libtool. We can work around
that dependency by copying the macro into our file that contains our
custom macros, acinclude.m4 if we would like.
I think we should do that, especially because it will reduce the size of
the generated
Alexei Volkov wrote:
Is it possible (in theory) to make asterisk server multiple sip endponts
configured with same sip credentials.
Of course it's possible (in theory). Asterisk is software, software can
be programmed to do anything people want it to do.
snip
If asterisk can support multiple
Luigi Rizzo wrote:
i.e. tell it what extra libs we need.
I modified the macro (which is our code, in acinclude.m4)
to support multiple instances and stop at the
first matching one e.g.
AST_EXT_LIB_CHECK([TONEZONE], [tonezone], [tone_zone_find],
[zaptel/tonezone.h], [${tonezone_extra}],
After many (many) hours of work, I have completed a significant upgrade
to our Subversion infrastructure.
First, our Subversion servers are now running Subversion 1.4.0, and all
the repositories have been upgraded to the new 1.4 higher-compression
format. This will results in faster checkouts,
Tzafrir Cohen wrote:
* 'make all' should not depend on the menuselect target.
Why not? We will decide which utilities to build based on the dependency
information found by the configure script, which menuselect then turns
into target information for the Makefile.
* menuselect should be run
Luigi Rizzo wrote:
i asked this specific change on the list, we discussed a bit on how
i thought it was useful. i did not see any explaination from you
(or others, if it matters) on why you consider this code harmful.
We don't reject things ONLY because they might be harmful. We also have
to
Luigi Rizzo wrote:
after this commit acloca.m4e went up from one line to over 6200:
http://svn.digium.com/view/asterisk/trunk/aclocal.m4?rev=46846view=log
surely there must be a more compact way to figure out whether we
are using gnu-ld or something else ?
This macro is what GNU libtool
SF Markus Elfring wrote:
I propose to use abnormal program termination consistently.
Can you see the relevance now?
Does my alternative description show the opportunities for the requested
corrections?
Sure, but it does nothing to improve the situation except to reiterate
that the code could
Luigi Rizzo wrote:
yes there is no fopencookie manpage, but the call is present e.g. on
the linux box where you gave me an account, and where i actually
tested that the code was correctly handling https.
You can try for yourself.
Note that fopencookie(), as well as funopen(), has no
Brian Candler wrote:
3. Revert to 1.2 behaviour (i.e. offer all codecs with media proxying, and
re-INVITE to set up native bridging afterwards). However, re-order the codec
list so that the codecs included in the incoming SIP INVITE appear before
the others. This might make it a bit more
Josh McAllister wrote:
I sense from the short tone of your response that you feel this to be
off topic here. Is there a better forum for me to start a dialogue about
this? I understand some of the questions could have been answered myself
with enough time spent sifting through code, but does
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