On Thu, Dec 27, 2007 at 02:33:22PM -, SVN commits to the Asterisk project
wrote:
> Author: russell
> Date: Thu Dec 27 08:33:21 2007
> New Revision: 94828
>
> URL: http://svn.digium.com/view/asterisk?view=rev&rev=94828
> Log:
> Change ast_translator_best_choice() to only pay attention to audio
On Wed, Dec 19, 2007 at 02:14:13PM -0800, Luigi Rizzo wrote:
> On Wed, Dec 19, 2007 at 10:36:34PM +0100, Michiel van Baak wrote:
> > On 14:26, Wed 19 Dec 07, Jason Parker wrote:
...
> > > This _SETVAR macro may not be working quite as you had intended. If you
> > > s
On Wed, Dec 19, 2007 at 10:36:34PM +0100, Michiel van Baak wrote:
> On 14:26, Wed 19 Dec 07, Jason Parker wrote:
> > SVN commits to the Digium repositories wrote:
> > > Author: rizzo
> > > Date: Tue Dec 18 04:24:58 2007
> > > New Revision: 93603
> > >
> > > URL: http://svn.digium.com/view/asterisk
On Sat, Dec 15, 2007 at 09:19:53PM +0200, Tzafrir Cohen wrote:
> On Sat, Dec 15, 2007 at 10:07:32AM -0800, Luigi Rizzo wrote:
...
> If the scanning is too slow, maybe it could be optimized. One way is to
> replace everything with a perl/python/whatever script. Such a script
> will
On Mon, Dec 10, 2007 at 06:36:30AM -0600, Kevin P. Fleming wrote:
> Luigi Rizzo wrote:
>
> > BTW there are three more places in this file where there is a ../ which
> > maybe
> > should become $(ASTTOPDIR) ?
> >
> > __embed_ldscript:
> >
On Thu, Nov 29, 2007 at 01:24:34PM -0600, Tilghman Lesher wrote:
> On Thursday 29 November 2007 11:46:07 Kevin P. Fleming wrote:
...
> > > +++ trunk/include/asterisk/lock.h Thu Nov 29 11:42:21 2007
> > > @@ -113,6 +113,8 @@
> > > { PTHREAD_MUTEX_INIT_VALUE, 0, { NULL }
On Mon, Nov 26, 2007 at 11:29:37AM -0600, Kevin P. Fleming wrote:
> SVN commits to the Asterisk project wrote:
>
> > -struct ast_hashtab *ast_hashtab_create(int initial_buckets,
> > - int (*compare)(const void *a, const void *b), /* a func to compare two
> > elements in the hash -- cannot be nu
On Wed, Nov 21, 2007 at 03:45:56PM -, SVN commits to the Asterisk project
wrote:
> Author: kpfleming
> Date: Wed Nov 21 09:45:56 2007
> New Revision: 89481
>
> URL: http://svn.digium.com/view/asterisk?view=rev&rev=89481
> Log:
> get this to actually compile...
interesting... another client f
On Wed, Nov 21, 2007 at 09:28:55AM +0100, Olle E Johansson wrote:
> >
> > Really, this is not an area where you can afford playing and putting
> > in
> > small patches to see how they fix one or the other problem.
> > The correctness of extension matching is something that people
> > really must
On Tue, Nov 20, 2007 at 01:28:34PM -0500, Simon Perreault wrote:
> On Tuesday 20 November 2007 13:20:49 Luigi Rizzo wrote:
> > it is actually a big burden. there are 145 different version of each of the
> > tools, some of which are incompatible with each other,
> > and
On Tue, Nov 20, 2007 at 08:33:28PM +0200, Tzafrir Cohen wrote:
> Hi, thanks for your quick reply,
>
> On Tue, Nov 20, 2007 at 10:14:02AM -0800, Luigi Rizzo wrote:
>
> > With this in mind, i think we have three type of headers in asterisk
> > (i use the current name/loca
On Tue, Nov 20, 2007 at 12:10:50PM -0600, critch wrote:
> On Tue, 2007-11-20 at 11:32 -0500, Simon Perreault wrote:
> > On Tuesday 20 November 2007 11:16:33 Tzafrir Cohen wrote:
> > > So you would want every user who isntalls Asterisk from SVN to have
> > > autoconf and automake? (what versions exa
On Tue, Nov 20, 2007 at 07:39:04PM +0200, Tzafrir Cohen wrote:
> Hi
>
> With all of Luigi's work on reducing the number of include files, we're
> still left with the need to include several files.
>
> Currently there are two include files directly under include/ -
> asterisk.h and jitterbuffer.h
On Tue, Nov 20, 2007 at 11:01:09AM -0500, Simon Perreault wrote:
> On Tuesday 20 November 2007 10:39:00 SVN commits to the Asterisk project
> wrote:
> > add an argument for extra headers to AC_EXT_LIB_CHECK,
> > and on passing simplify the code.
> > Too bad that every time we need to regenerate co
is there any work going on for implementing struct ast_channel with astobj2?
I am running in a problem with the video extensions that i think
already came out in some other channel driver, and would be completely
solved implementing ast_channel using refcounted structures.
In this specific case,
On Fri, Nov 09, 2007 at 09:40:35AM -0700, Steve Murphy wrote:
> OK, I just committed the beast (see below). I did some last minute
> code review, cleanup and some work to improve the cid matching code.
see my followup to the commit message - i really encourage
you to consider that change in integr
On Fri, Nov 09, 2007 at 04:00:23PM -, SVN commits to the Asterisk project
wrote:
> Author: murf
> Date: Fri Nov 9 10:00:22 2007
> New Revision: 89129
>
> URL: http://svn.digium.com/view/asterisk?view=rev&rev=89129
> Log:
> This is the perhaps the biggest, boldest, most daring change ...
I h
On Tue, Nov 06, 2007 at 10:39:52AM -0600, Russell Bryant wrote:
> Luigi Rizzo wrote:
> > I have an ugly fix, below - store the relevant values from
> > module_list in a static variable not subject to constructors,
> > and restore module_list from that variable near
On Tue, Nov 06, 2007 at 09:29:24AM -0600, Kevin P. Fleming wrote:
> Luigi Rizzo wrote:
>
> > I was under the impression that they are embedded object coming from C++
> > sources, so the build path should be .cc -> .oo -> .eoo , but
> > if you look at Makefile.moddi
I found the bug that causes module embedding not to work on FreeBSD
(and maybe on other architectures too).
Module embedding is implemented using some linker magic that
causes ast_module_register() to be called for each of the
embedded objects.
This in turn adds the entry to a list, module_list, i
On Tue, Nov 06, 2007 at 04:08:33PM +0100, Michael Iedema wrote:
> On 11/6/07, Luigi Rizzo <[EMAIL PROTECTED]> wrote:
> >
> > ok it works on linux but not on FreeBSD. Need to check why - any
> > documentation or suggestion on what mechanism is used to make dlopen()
>
On Tue, Nov 06, 2007 at 05:42:33AM -0800, Luigi Rizzo wrote:
> On Tue, Nov 06, 2007 at 06:37:18AM -0600, Kevin P. Fleming wrote:
> > Luigi Rizzo wrote:
> >
> > > Is there any trick i should use here ?
> >
> > Unless the loader or other bits have been broken re
I have a longer message coming up, but just noticed that stringfields.h
says that ast_string_field_free_all() is enough to release storage:
When the structure instance is no longer needed, the fields
and their storage pool must be freed:
\code
ast_string_field_free_all(samp
On Tue, Oct 30, 2007 at 10:26:02AM -0500, Kevin P. Fleming wrote:
> Luigi Rizzo wrote:
>
> > ok thanks for the clarification.
> > The absence of sequence number makes it slightly difficult to detect
> > missing frames though...
>
> Not really... sequence numbers
On Tue, Oct 30, 2007 at 10:13:56AM -0500, Kevin P. Fleming wrote:
> SVN commits to the Asterisk project wrote:
> > Author: rizzo
> > Date: Tue Oct 30 09:57:19 2007
> > New Revision: 87533
> >
> > URL: http://svn.digium.com/view/asterisk?view=rev&rev=87533
> > Log:
> > remove some useless arguments
On Tue, Oct 30, 2007 at 08:42:45AM -0500, Kevin P. Fleming wrote:
> Luigi Rizzo wrote:
>
> > On the other hand, you wouldn't expect the compiler to fail so badly.
>
> I really do not understand this comment at all. First, all software has
> bugs, compilers included
On Mon, Oct 22, 2007 at 05:29:40PM -0500, Russell Bryant wrote:
...
> I have been thinking about this, and I do not think that an API/ABI freeze is
> something that I would like to do for Asterisk 1.6 at this point.
>
> As has been pointed out in other parts of this thread, trunk should always be
On Tue, Oct 16, 2007 at 08:12:48AM +0200, Olle E Johansson wrote:
>
> 14 okt 2007 kl. 21.47 skrev Luigi Rizzo:
...
> > Anyways, what would be wrong with the obvious solution of notify
> > the caller's channel driver once the remote party reports supported
> > formats
On Mon, Oct 01, 2007 at 01:47:26PM -0400, Simon Perreault wrote:
> On Monday 01 October 2007 13:30:46 Jason Parker wrote:
> > Just a thought, but would it maybe be useful to have something like
> > AST_LIST_COMBINE, to add the contents of list2 to list1?
>
> Well, if I understand correctly, the pr
i just realised "the hard way" that AST_LIST_INSERT_TAIL can only
append a single element to a list, and it is not good for appending
two lists. The macro also assumes that the element has the 'tail'
pointer properly initialized.
I am not 100% sure that all places in asterisk where AST_LIST_INSERT
The crash that I and others were seeing in in chan_iax happens
early in the load process, presumably while loading the config file.
The stack trace is this one:
#0 user_hash_cb (obj=0xbfbfef3c, flags=8) at
/usr/ports/net/asterisk-test/work/asterisk-devel-1.4/include/asterisk/strings.h
:677
#1 0
On Thu, Sep 20, 2007 at 09:21:29PM -, SVN commits to the Asterisk project
wrote:
> Author: mmichelson
> Date: Thu Sep 20 16:21:28 2007
> New Revision: 83350
>
> URL: http://svn.digium.com/view/asterisk?view=rev&rev=83350
> Log:
> Merging changes from queue_refcount_trunk into trunk. Refcounte
On Thu, Sep 20, 2007 at 11:32:04AM -0500, Russell Bryant wrote:
> Stephen Davies wrote:
...
> > I must say that I tried to update my live service box last Tuesday to
> > trunk rev 81432 - with your new astobj2 stuff in chan_iax2. I don't
> > know if I messed up the upgrade, but it just crashed and
On Tue, Sep 18, 2007 at 11:06:12AM -0500, Jason Parker wrote:
> SVN commits to the Digium repositories wrote:
> > Author: rizzo
> > Date: Tue Sep 18 10:45:22 2007
> > New Revision: 82753
> >
> > URL: http://svn.digium.com/view/asterisk?view=rev&rev=82753
> > Log:
> > lots of cleanup of this module
Hi,
i have reached a hopefully reasonable state for the code in
http://svn.digium.com/view/asterisk/team/rizzo/video_v2/
so it would be good if someone could give it a try and send feedback.
In a nutshell, this branch lets you send and receive (and display)
H263+ video with chan_oss (tes
On Mon, Sep 17, 2007 at 11:28:22PM +0200, Tzafrir Cohen wrote:
> On Mon, Sep 17, 2007 at 02:03:29PM -0700, Luigi Rizzo wrote:
> > On Mon, Sep 17, 2007 at 03:51:35PM -0500, Tilghman Lesher wrote:
> > > On Monday 17 September 2007 15:03, Luigi Rizzo wrote:
> > > > so
sorry if the question is trivial...
while adding videosupport to chan_oss/chan_alsa, which you can find at
http://svn.digium.com/view/asterisk/team/rizzo/video_v2/
I just realised that the various video sources that I am using
(webcam using the 'gspca' driver, or X11 grabber) do not support
On Thu, Sep 13, 2007 at 04:30:50PM -, SVN commits to the Asterisk project
wrote:
> Author: russell
> Date: Thu Sep 13 11:30:50 2007
> New Revision: 82328
>
> URL: http://svn.digium.com/view/asterisk?view=rev&rev=82328
> Log:
> * constify some args to manager functions
> * add two new function
On Sun, Sep 09, 2007 at 01:09:07PM +1000, David Bowerman wrote:
> Hi all,
>
> Ive been working on a project to add i/o controller (home automation)
> support to asterisk so you can do things like switch on and off
> devices and sense the state of them from within the dialplan.
>
> Its mainly base
On Sun, Sep 09, 2007 at 02:35:18AM -, SVN commits to the Asterisk project
wrote:
> Author: tilghman
> Date: Sat Sep 8 21:35:18 2007
> New Revision: 82028
>
> URL: http://svn.digium.com/view/asterisk?view=rev&rev=82028
> Log:
> Fix inline compiles on really old compilers (who uses gcc 2.7 any
On Wed, Sep 05, 2007 at 08:02:39PM +1000, Devraj Mukherjee wrote:
> Thanks Luigi.
>
> Any links/docs on how to configure this?
look in manager.conf and http.conf it's all documented there
(and trivial, indeed)
luigi
___
--Bandwidth and Colocation Prov
On Wed, Sep 05, 2007 at 05:13:55PM +1000, Devraj Mukherjee wrote:
> Hi everyone,
>
> I just found this post when searching for AMI over SSL
> http://bugs.digium.com/view.php?id=6812
>
> Does anyone know if this ever made it to the code base?
not this one but a different implementation is in trun
On Wed, Sep 05, 2007 at 03:44:32AM -, SVN commits to the Asterisk project
wrote:
> Author: russell
> Date: Tue Sep 4 22:44:31 2007
> New Revision: 81482
>
> URL: http://svn.digium.com/view/asterisk?view=rev&rev=81482
> Log:
> i am ok without the boost setting, you can use file\'s func_volume
On Wed, Sep 05, 2007 at 03:52:56AM -, SVN commits to the Asterisk project
wrote:
> Author: russell
> Date: Tue Sep 4 22:52:56 2007
> New Revision: 81490
>
> URL: http://svn.digium.com/view/asterisk?view=rev&rev=81490
> Log:
> Add a "console flash" CLI command, knocking another line off of th
On Wed, Aug 29, 2007 at 01:34:41PM -0500, Russell Bryant wrote:
> Luigi Rizzo wrote:
> > As far as i remember pointer arithmetic on "void *" is a gnu
> > extension not guaranteed to work on all compilers.
> > You should use char * to be standard compliant (even bette
On Wed, Aug 29, 2007 at 04:07:36PM -, SVN commits to the Asterisk project
wrote:
> Author: file
> Date: Wed Aug 29 11:07:35 2007
> New Revision: 81345
>
> URL: http://svn.digium.com/view/asterisk?view=rev&rev=81345
> Log:
> This concludes bringing trunk back to a working state.
>
> Modified:
On Thu, Jul 26, 2007 at 01:10:49PM -, SVN commits to the Asterisk project
wrote:
> Author: russell
> Date: Thu Jul 26 08:10:49 2007
> New Revision: 77266
>
> URL: http://svn.digium.com/view/asterisk?view=rev&rev=77266
> Log:
> Add a link to the list of assigned RTP payload types for convenien
On Tue, Jul 24, 2007 at 10:41:55AM +0200, Gregory Nietsky wrote:
>
>
> Hi there all I have written a stun client that fits into rtp at the moment
> it basically completes a RFC stun discovery using the socket of the rtp
> struct.
>
> This allows the manipulation of rtp traffic for nat traversal.
On Wed, Jul 18, 2007 at 08:00:23PM -, SVN commits to the Asterisk project
wrote:
> Author: file
> Date: Wed Jul 18 15:00:23 2007
> New Revision: 75712
>
> URL: http://svn.digium.com/view/asterisk?view=rev&rev=75712
> Log:
> Backport GCC 4.2 fixes. Without these Asterisk won't build under devm
On Wed, Jul 18, 2007 at 01:22:47PM -0500, Kevin P. Fleming wrote:
> Steve Murphy wrote:
>
> > But I agree with you-- the principle "minimal change" seems better; what
> > if I
> > created an analog to ast_flags, like ast_flags64, and a few APP_ARG
> > alternates that would use 64-bit flags, that a
On Wed, Jul 18, 2007 at 10:11:54AM -0700, John Todd wrote:
> At 7:37 AM -0500 2007/7/18, Tilghman Lesher wrote:
> >On Wednesday 18 July 2007, Luigi Rizzo wrote:
> >> I would like to merge into trunk at least (and later, possibly
> >> also into 1.4 if considered us
On Mon, Jul 16, 2007 at 07:34:26AM -0700, Luigi Rizzo wrote:
> On Mon, Jul 16, 2007 at 04:13:55PM +0200, Klaus Darilion wrote:
> > Hi!
> >
> > Are somewhere common functions for string/integer conversion defined?
re-reading the subject, it seems that you asked for int
On Mon, Jul 16, 2007 at 04:13:55PM +0200, Klaus Darilion wrote:
> Hi!
>
> Are somewhere common functions for string/integer conversion defined?
i don't think there are. As a matter of fact i am working right now
on a common function to convert argument into int/sockaddr/port
numbers, with optiona
On Wed, Mar 28, 2007 at 07:33:23AM +0200, Olle E Johansson wrote:
>
> 28 mar 2007 kl. 01.16 skrev Nicholas Campion:
>
> > You could create another table in the database that would hold the
> > last write times of these tables. Create a separate table with the
> > columns "table name" and "mo
On Tue, Mar 27, 2007 at 02:35:05PM -0600, Steve Murphy wrote:
> OK, I just closed 9037 after a conversation with kpfleming.
> Here is the situation:
>
> 9037's reporter is complaining that with a large dialplan, in order to
> look up an extension, and do the pattern matching, asterisk pulls every
As discussed with kevin, the following patch for trunk
largely simplifies and improves the detection of zaptel features
in configure.ac - it retains the current semantics for linux, and
adds the ability to put overrides for other platform where an
up-to-date version of the zaptel drivers does not e
On Wed, Dec 27, 2006 at 10:14:34PM -, [EMAIL PROTECTED] wrote:
> Author: kpfleming
> Date: Wed Dec 27 16:14:33 2006
> New Revision: 49008
>
> URL: http://svn.digium.com/view/asterisk?view=rev&rev=49008
> Log:
> Merged revisions 49006 via svnmerge from
> https://origsvn.digium.com/svn/asterisk
I find this svnmerge feature exceedingly annoying:
> svnmerge block -r 48566 -P branch-1.4-merged -B branch-1.4-blocked
svnmerge: "." has local modifications; it must be clean
Given that I do need local modifications to build stuff on my
(non-linux) systems, this means that to run svnmerg
luigi
> Mog
> - Original Message -
> From: Luigi Rizzo <[EMAIL PROTECTED]>
> To: Asterisk Developers Mailing List
> Sent: Tuesday, December 12, 2006 4:56:25 PM GMT-0600 US/Central
> Subject: [asterisk-dev] gratuitous change in rev.48416 ?
>
> i
i don't understand the change in rev.48416 :
it seems that ZT_TCOP_RELEASE is not used here, and if
anything, it's codecs/codec_zap.c that fails to build,
not this one, so why make the compiler fail here and
not in the other module ?
I know that you are fond of the latest and greatest
versions o
i am pretty sure the following is a bug, but can someone confirm it ?
When making an outgoing call, i noticed that even if the caller
hangs up before the callee answers, asterisk does not sent a CANCEL.
It took a bit to reproduce it, but now i know that it happens when
the callee sends back a 401
On Fri, Dec 08, 2006 at 05:48:10PM +0100, Martin Vít wrote:
> Jared Smith wrote:
> > On 12/7/06, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> >> > Something that I find bothering when I try to debug Asterisk is
> >> that it
> >> > deamonizes before most errors can occour. It will fork into backgroun
On Wed, Nov 29, 2006 at 05:08:19AM -, [EMAIL PROTECTED] wrote:
> Author: russell
> Date: Tue Nov 28 23:08:19 2006
> New Revision: 48103
>
> URL: http://svn.digium.com/view/asterisk?view=rev&rev=48103
> Log:
> Remove an XXX command suggesting that this truncation should not be
> conditional,
>
On Sun, Nov 26, 2006 at 06:55:34AM -, [EMAIL PROTECTED] wrote:
> Author: russell
> Date: Sun Nov 26 00:55:33 2006
> New Revision: 48019
>
> URL: http://svn.digium.com/view/asterisk?view=rev&rev=48019
> Log:
> - Add some comments on thread storage with a brief explanation of what it is
> as w
could someone explain how the code in threadstorage.h is used
and what is it supposed to do ?
I am not sure i follow the code too well because it is hidden
behind several macros, nor i could find in the code any comment
on what this code is supposed to provide.
Browsing through the asterisk sourc
On Wed, Nov 15, 2006 at 02:10:57PM -0600, Kevin P. Fleming wrote:
> 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 su
On Wed, Nov 15, 2006 at 09:47:22PM +, Tony Mountifield wrote:
> In article <[EMAIL PROTECTED]>,
> Tilghman Lesher <[EMAIL PROTECTED]> wrote:
> > On Wednesday 15 November 2006 12:32, Luigi Rizzo wrote:
> > > > I think they make this part of the code a lot
On Wed, Nov 15, 2006 at 06:24:28PM -, [EMAIL PROTECTED] wrote:
> Author: rizzo
> Date: Wed Nov 15 12:24:28 2006
> New Revision: 47681
>
> URL: http://svn.digium.com/view/asterisk?view=rev&rev=47681
> Log:
> start using the macros already present in chan_oss.c to parse
> the configuration file.
On Fri, Nov 10, 2006 at 06:10:23PM -0600, JR Richardson wrote:
>
> >
> > I'm having a strage problem related to the way Asterisk is matching
> > SIP peers on inbound calls to DIDs (not REGISTERed extensions). I
>
> I just had this same issue as well, drove me crazy. I have a different
> setup
On Thu, Nov 09, 2006 at 03:25:36PM -0600, Kevin P. Fleming wrote:
> 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
On Thu, Nov 09, 2006 at 01:48:45PM -0600, Kevin P. Fleming wrote:
> 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 proble
any ideas on the second part, __sip_ack() leaking memory ?
cheers
luigi
On Thu, Nov 09, 2006 at 04:19:05PM -, [EMAIL PROTECTED] wrote:
> Author: rizzo
> Date: Thu Nov 9 10:19:05 2006
> New Revision: 47373
>
> URL: http://svn.digium.com/view/asterisk?view=rev&rev=47373
> Log:
> rename the "o
On Wed, Nov 08, 2006 at 09:53:55AM -0600, Kevin P. Fleming wrote:
> 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 programme
On Wed, Nov 08, 2006 at 09:46:39AM -0600, Kevin P. Fleming wrote:
> 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 su
don't know if it is just me or it is a common thing,
but i am receiving replies to the -dev list with a
delay between 30 and 60 minutes (i can tell because i see
them on the web interface, and my own postings appear there
right away).
Do others experience the same delay ?
cheers
l
On Mon, Nov 06, 2006 at 03:21:59PM -0600, Kevin P. Fleming wrote:
> Luigi Rizzo wrote:
>
> > iksemel is third party software that uses whatever its authors
> > decided to use, and i don't think anyone has the interest of rewriting
> > it to use openssl.
>
> A
On Mon, Nov 06, 2006 at 03:16:45PM -0600, Kevin P. Fleming wrote:
> 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
we have this block in configure.ac:
AST_EXT_LIB_CHECK([IKSEMEL], [iksemel], [iks_start_sasl], [iksemel.h])
if test "${PBX_IKSEMEL}" = 1; then
AST_EXT_LIB_CHECK([GNUTLS], [gnutls], [gnutls_bye])
if test "${PBX_GNUTLS}" = 1; then
IKSEMEL_LIB="${I
On Sat, Nov 04, 2006 at 11:59:41PM +0100, Olle E Johansson wrote:
> Friends,
> We're in SSL/TLS hell and need a strategy to get to SSL/TLS heaven,
> if it exists ;-)
>
> Currently we have many different implementations:
>
> * John Todd's SSL for manager API in the bug tracker (OpenSSL)
> * The
On Tue, Oct 31, 2006 at 10:09:19AM +0200, Alexandr Olekhnovich wrote:
> Hello all.
> Can anybody explain me, what the next definitions mean in asterisk modules
> and when I can use them?
> STANDARD_LOCAL_USER
> LOCAL_USER_DECL
> STANDARD_HANGUP_LOCALUSER
> Thank you.
they are a relic from the past
T286 does this
(addresses replaced with x.x.x.x). As you see, the Contact: has
nothing useful, and the name is in the From: field.
<--- SIP read from x.x.x.x:29848 --->
REGISTER sip:x.x.x.x SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:29848;branch=z9hG4bK4b52818f0289e5fa
From: "Luigi Riz
trings containing a '\0'
are handled - basically, get_input() uses \r\n as an end-of-line
marker, but other code takes a \0 as end of line
cheers
luigi
> Julian
>
> Luigi Rizzo wrote:
> > On Wed, Oct 25, 2006 at 05:36:24PM +0100, Julian Lyndon-Smith wrote:
> >>
On Thu, Oct 26, 2006 at 08:10:35AM +0200, Johansson Olle E wrote:
> >> Log:
> >> Somewhat ugly code to try to fix issue #7608.
> >> Since the problem was not very well defined, the fix is a bit
> >> fuzzy too...
> >> Thanks to Luigi for accidentally spotting the possible problem!
> >>
> >> M
On Wed, Oct 25, 2006 at 07:16:11PM -, [EMAIL PROTECTED] wrote:
> Author: oej
> Date: Wed Oct 25 14:16:10 2006
> New Revision: 46252
>
> URL: http://svn.digium.com/view/asterisk?rev=46252&view=rev
> Log:
> Somewhat ugly code to try to fix issue #7608.
> Since the problem was not very well defi
On Mon, Oct 23, 2006 at 05:41:00PM -0500, Kevin P. Fleming wrote:
> Luigi Rizzo wrote:
> > well, it was trivial :)
> > see the team/rizzo/astobj2 branch, file main/http.c
>
> 'man fopencookie' on my Debian unstable system produces an error. Also,
> there is no
On Sun, Oct 22, 2006 at 09:25:57AM -0700, Luigi Rizzo wrote:
> On Sun, Oct 22, 2006 at 03:09:46PM +, Tony Mountifield wrote:
> > In article <[EMAIL PROTECTED]>,
> > Luigi Rizzo <[EMAIL PROTECTED]> wrote:
> > > as the subject says...
> > >
> >
On Sun, Oct 22, 2006 at 05:35:58PM +0100, Tim Panton wrote:
>
> On 22 Oct 2006, at 17:02, Luigi Rizzo wrote:
...
> >> I'd like to hear a discussion of where the 'http/manager' facilities
> >> are going.
> >> At the moment they are in a limb
On Sun, Oct 22, 2006 at 03:09:46PM +, Tony Mountifield wrote:
> In article <[EMAIL PROTECTED]>,
> Luigi Rizzo <[EMAIL PROTECTED]> wrote:
> > as the subject says...
> >
> > read below for the details - if someone autoconf expert
> > can suggest ho
On Sun, Oct 22, 2006 at 04:53:52PM +0100, Tim Panton wrote:
>
> On 22 Oct 2006, at 06:18, Joshua Colp wrote:
>
> > Greetings and Salutations Folks!
> >
> > As you all probably know we are having a Developer Summit at
> > Astricon on the fast approaching Tuesday of next week. Participants
> >
Best Regards,
>
> King
>
> -ì©l¶l¥ó-
> ±H¥óªÌ: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] ¥N²z Luigi Rizzo
> ±H¥ó¤é´Á: Thursday, 19 October, 2006 6:31
> ¦¬¥óªÌ: Asterisk Developers Mailing List
> ¥D¦®: [asterisk-dev] sip authentication again
>
> subject
subject changed to raise attention :)
taking kevin's ideal model below:
On Mon, Oct 16, 2006 at 12:27:21PM -0500, Kevin P. Fleming wrote:
...
> If the service they are asking to INVITE is open to the world,
> then the INVITE goes through unchallenged and you don't know who
> is connecting to you.
On Mon, Oct 16, 2006 at 01:20:21PM -, [EMAIL PROTECTED] wrote:
> Author: oej
> Date: Mon Oct 16 08:20:21 2006
> New Revision: 45209
>
> URL: http://svn.digium.com/view/asterisk?rev=45209&view=rev
> Log:
> When adding new functions, please add a forward declaration.
> I *know* it is not require
according to the RFC and a bit of googling on mailing lists,
zero (0) seems to be a legitimate value for SIP CSeq numbers,
so unless there are objections or other reasons i would like
to remove the special handling of 0 for ocseq/icseq in chan_sip.c
Otherwise, if we need to keep it, i would like a
't check the user
objects (or realtime) */
user = find_user(of, 1);
On Mon, Oct 09, 2006 at 02:48:51PM -0700, Luigi Rizzo wrote:
> Hi,
> i am experiencing the following and i am not sure if it is caused
> by a local misconfiguration, a fundamental problem with
Hi,
i am experiencing the following and i am not sure if it is caused
by a local misconfiguration, a fundamental problem with SIP, or
a problem in the way asterisk matches INVITEs against directly
connected SIP devices.
I have asterisk connected to a SIP provider, and to some SIP phones
(type=frie
On Thu, Oct 05, 2006 at 03:34:02PM +0200, Tzafrir Cohen wrote:
> On Thu, Oct 05, 2006 at 04:09:29AM -0700, Luigi Rizzo wrote:
>
> > The problem is that platform-specific _defaults_ should be provided
> > by the tool (which should aware of platform issues) not by the
> >
On Thu, Oct 05, 2006 at 11:11:00AM +0200, Tzafrir Cohen wrote:
> On Thu, Oct 05, 2006 at 12:15:33AM -0700, Luigi Rizzo wrote:
...
> > Besides, the "failures" are often subtle - it's not that configure
> > fails completely, it just gives you a suboptimal build becaus
On Wed, Jul 12, 2006 at 02:21:49AM -0400, Brian Capouch wrote:
> I suspect I may need to upgrade my gcc, but I haven't seen mention of
> such. I try to watch all the postings about the development tree:
>
> make[1]: Leaving directory `/usr/src/asterisk/agi'
> make -C db1-ast libdb1.a
> make[1]:
On Fri, Apr 21, 2006 at 08:48:48AM -0500, Tilghman Lesher wrote:
> On Friday 21 April 2006 06:09, Julian Lyndon-Smith wrote:
> > FWIW, I would say that "" is the best.
>
> Actually, that should be "Unknown <>", since the number field
> should be empty.
We are talking about individual fields here
luigi
On Tue, Apr 18, 2006 at 09:50:59AM -0700, Luigi Rizzo wrote:
> hi,
> browsing throughout the code, i see that some places map
> a NULL value in the struct ast_callerid fields to ""
> or similar string, but the empty string is left alone.
>
> However, i am
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