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.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experie
rations that increment the
> ref count, with operations that are meant to decrement the refcount.
Umm... s/expanation/explanation/ :-)
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
n that ztscan can use; now
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
ceive before it can send you audio, as already commented on, is very
confusing and easy to misunderstand.
I can certainly agree though that if we haven't been told the other end
will accept audio, that we should drop any audio we receive from them.
--
Kevin P. Fleming
Director of Software Techno
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.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk
phone and the phone is free to send it.
The phone's response without SDP means Asterisk should not send audio
*to* the phone, but it does not impact Asterisk receiving audio.
--
Kevin P. Fleming
Director of Software Technologies
Dig
e none of the
low-level drivers currently support this flag.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
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aste
t
you control the state of via whatever logic you choose and then let
app_queue monitor those to determine the availability of queue members.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
__
tom prefix you'd like to use.
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Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
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T
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.
--
Kevin P. Fleming
Director of Software Technol
hat this morning. I just 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.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc.
he Brazilian 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.
--
modify them when
we're really, really sure we should :-)
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
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$(patsubst %,$(SUBDIR)/%,$(filter %.eo,$^)); do 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 link
irectory entry instead. This will make it
very easy for an admin to change the location for one item.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
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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?
--
Ke
ase model, it *could* be
added during 1.6's lifetime, rather than having to wait for 1.8.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
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This makes no sense; once 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?
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Ge
inition of 'consume'? The application you are writing
is the one 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.
--
Kevin
tended? They should be put back, and using Doxygen syntax as the
coding guidelines mandate.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
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Johansson Olle E wrote:
> Join me in congratulations to Philippe for this election!
Fantastic! Glad to hear it Philippe!
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
--
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.
--
fferentiate them ?
Sorry, I 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.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The
ile-level 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.
--
Kev
t; .eoo , and besides,
> later .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.
--
y time period is missing media.
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Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
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To U
nd merge them for you.
You have commit access to trunk, please make the changes directly there
if you feel they are worth making.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
__
7;t more difficult to maintain, then feel free to contribute them.
However, requiring developers to change the way they use the macros, the
application to use more memory or increase our maintenance burden, all
to support a six year old compiler with dozens of known bugs, seems like
a fruitless pur
hould not be
called directly.
Another possibility would be to move the definition of the
dialed_interface_info structure into global_datastores.c, and leaving
just an 'extern' declaration of it in this header file. This would allow
these two functions to then be stati
support into
their own packages without having to accept the new features or other
more invasive changes that are present in the newer releases.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
hanges. This new release
model means that they could have to make 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.
--
Kevin P. Fleming
Director of S
the
parking lot will still be populated by contractor vehicles, moving vans
and others who don't have anything to do with Digium directly...
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
27;d prefer 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.
--
Kevin P. Fleming
Director of S
s just working in that code yesterday 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).
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "
or we need to simplify the code in app_queue and just populate
->membername with a copy of ->interface if no member name is provided
when the member is added to the queue.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
sking for here. I don't see why we shouldn't do the same
thing in chan_iax2.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
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s not a 'feature request',
it's 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 :-)
--
Kevin P. Fleming
Director of Softw
re.
2) I've removed this entirely 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.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc.
being
temporary... in that case, never mind :-)
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
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asterisk-de
when this change is made to the
server. We will begin using the downloads.digium.com name in all our
announcements, security advisories, documentation and other places
immediately, and encourage the community to do so as well.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - &quo
ago.)
Well... 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.
--
Kevin P. Fleming
Director of Software Te
sk that uses locks depends 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.
--
Kevin P. Fleming
Director of Softw
sting users don't have to be
modified.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
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asterisk-dev mailing
nce. I don't see any reason to install it by default.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
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in trunk and rename sethdlc-new to sethdlc.
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Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
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To
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 :-)
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience
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.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience
e 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 :-)
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuin
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 bri
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
cro
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
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 th
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 some
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 ma
[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=diff&rev=60325&r1=60324&r2=60325
> ==
> --- branches/1.4/formats/f
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=rev&rev=2334
>> Log:
>> remove pointless file ID lines
>
> Why are the
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:
> it would have been nice to know a problem existed in chan_sip (on the
> website, without having to ask or searching the commits list) and great
> if a advisory was posted to one or all of the popular security mailing
> lists.
The fixed versions of Asterisk were posted wi
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
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
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 arch
ity 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
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
>
> Sh
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
> /usr/inc
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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To UNSUBSCRIBE or update opti
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 uniquei
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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[EMAIL PROTECTED] wrote:
> Author: file
> Date: Fri Jan 12 21:26:04 2007
> New Revision: 50676
>
> URL: http://svn.digium.com/view/asterisk?view=rev&rev=50676
> Log:
> Merge in trunkrealloc option for chan_iax2. (issue #8267 reported by
> marcodmb, branch by anthonyl)
Wouldn't it be clearer (and
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
> automaticall
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 pick
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=rev&rev=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?
Side note: PostgreSQL is one of the few packages that doesn't just a
simp
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
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 a
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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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 proces
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 th
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
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.
_
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; the
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
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
> r
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
[EMAIL PROTECTED] wrote:
> Author: rizzo
> Date: Wed Nov 15 08:11:28 2006
> New Revision: 47652
>
> URL: http://svn.digium.com/view/asterisk?view=rev&rev=47652
> Log:
> update the internal cli api following comments from kevin.
> This change basically simplifies the interface of the
> new-style ha
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 wrot
Luigi Rizzo wrote:
> any ideas on the second part, __sip_ack() leaking memory ?
Does running with MALLOC_DEBUG actually show memory being leaked? It
would be useful to know if there is an actual problem, as opposed to a
hypothetical problem that could take a long time to disprove.
Klaus Darilion wrote:
> Debian and openssl is not that easy. E.g. the openser shipped with
> debian has TLS disabled as it uses openssl and there are some conflicts
> with the GPL if openser is shipped as part of the Linux distribution.
That may be because the OpenSER license does not grant specif
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 i
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_ext
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.
> If asterisk can support multiple s
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
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 gnu
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 re-
[EMAIL PROTECTED] wrote:
> Author: file
> Date: Tue Nov 7 23:08:31 2006
> New Revision: 1570
>
> URL: http://svn.digium.com/view/zaptel?rev=1570&view=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
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=46846&view=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 lib
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