chitectures?
I believe if you are building for binary portability you need to disable
BUILD_NATIVE.
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packages?
I logged onto gerrit.asterisk.org, but was unable to find such a feature
(hopefully just because of my unfamiliarity).
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strange or by Asterisk.
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In article 53695c95.9060...@szeto.ca,
Daniel McFarlane dan...@szeto.ca wrote:
Hi Tony,
Hi Daniel,
See my answers inline.
On 05/06/2014 12:11 PM, Tony Mountifield wrote:
In article 5368ed8b.4080...@szeto.ca,
Daniel McFarlane dan...@szeto.ca wrote:
Hi All,
I've been working
asked on the asterisk-users list.
Yup, next limit you'll hit is dahdi pseudo channels, which is 512.
Is that limit configurable or fixed?
Tony
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While waiting for any replies to my first posting from the experts, I have
been doing some more studying of the usage of CHECK_BLOCKING() and the
flag AST_FLAG_BLOCKING. See below:
Tony Mountifield [EMAIL PROTECTED] wrote:
Yesterday I ported the MeetMe 'F' option (pass through DTMF) back to 1.2
to be sent to the
AGI's stdout, with the AGI response on stdin getting pushed back out as
the AMI response. Together with an AMI event at the start containing the
AGI environment, and presumably an AMI command to tell the AGI to return
to the dialplan.
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whatever it is used for be
achieved in some more compatible way?
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/Save dialog. This is much less convenient for browsing.
Is there any way to go back to the old behaiour?
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specific question:
You said multichannel bridging uses zaptel pseudo channels. If the call
is from a real Zap channel, does it bridge it directly, like Meetme does,
without using a pseudo?
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to conf_run().
Unless there's more to it than I thought. I haven't studied the SLA code
yet, since all I need is standard conferencing functionality. I always
wondered why SLA wasn't a separate module...
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be called from the 'h' extension, which will delete the file named
in the channel variable. These commands could be MeetmeNameRecord and
MeetmeNameDiscard, or perhaps just MeetmeName with an argument to specify
which operation should be performed.
Any comments?
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In article [EMAIL PROTECTED],
Ast Exp [EMAIL PROTECTED] wrote:
On 3/22/07, Tony Mountifield [EMAIL PROTECTED] wrote:
Look at the UserEvent dialplan application, e.g.
exten = _X.,n,UserEvent(SIP|SIPHdr: ${YOURSIPHEADER})
You would probably put this just before your Dial() statement
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to
be executed?
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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?
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In article [EMAIL PROTECTED],
Kevin P. Fleming [EMAIL PROTECTED] wrote:
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
are not as critical as bugs and which will receive second priority
in all cases.
If this policy has changed (and I'm not at all convinced it should), then
the guidelines ought to reflect this.
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suspected to reside in AMI.
Last Activity: 11/09
I'm not certain, but I suspect it could be a usage problem, trying to
use an Agent channel directly in a way it shouldn't be. I've just posted
a note to this bug suggesting an alternative approach.
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it
Thank you very much,
Phung
You can find all the information you need in the Wiki at
http://www.voip-info.org/wiki/view/Asterisk+manager+API
and in the file doc/manager.txt in the Asterisk distribution.
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Play
. I'd almost compare it to using macros for the sake of using
macros, not for any legitimate benefit.
I agree. When trying to understand code with lots of macros, I spend
most of my time looking back at the macro definitions to understand what
is actually going on!
Cheers
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In article [EMAIL PROTECTED],
Matthew Fredrickson [EMAIL PROTECTED] wrote:
On Nov 9, 2006, at 8:51 AM, Tzafrir Cohen wrote:
On Thu, Nov 09, 2006 at 01:27:32PM +, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Nov 09, 2006 at 12
In article [EMAIL PROTECTED],
Tilghman Lesher [EMAIL PROTECTED] wrote:
On Thursday 09 November 2006 06:10, Tony Mountifield wrote:
Is it possible from within a function to get a backtrace of where it was
called from? I would do this if I determined I was closing an unexpected
fd. Or perhaps
of `-'
chan_sip.c:1293: warning: comparison of distinct pointer types lacks a cast
chan_sip.c:1293: error: invalid type argument of `-'
Haven't tried it on 1.4 or trunk, but my first thought is: does ast_set_flag
exist in 1.2?
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that it has already locked, it will succeed, and the lock count
of the mutex will be incremented. When it unlocks the mutex, the lock
count will be decremented. When this count reaches zero, the mutex will
really be unlocked and available for another process to lock.
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in the right direction, I would be very
appreciative!
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is potentially a useful feature, but its implementation
shouldn't depend on a largely non-Linux library function! Since the vast
majority of Asterisk systems are Linux.
I'd be wary of code being committed to SVN that had only been tested on
BSD, and not on Linux.
Cheers
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() on
the relevant channels.
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changed due to the
different sizes of unsigned int and ast_group_t.
If you use an old chan_oh323, you will get all sorts of weird behaviour
because data in the struct isn't where chan_oh323 thinks it should be.
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Play
);
rtp-rtcp-us.sin_addr = addr;
}
Is there a reason not to?
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In article [EMAIL PROTECTED],
Kevin P. Fleming [EMAIL PROTECTED] wrote:
- Tony Mountifield [EMAIL PROTECTED] wrote:
What part of the code should be responsible for issuing ast_hangup()
on the
original channel B (seq .3)?
This should be handled by ast_do_masquerade(). However
that happens on the old channels. That is the event
that comes through the Manager API when zombie channels get discarded.
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cannot be for certain. The biggie so
far is log rotation. It can cause the system to bomb.
Joseph,
For the log rotation issue, please see http://bugs.digium.com/view.php?id=7195
and try one of the patches there.
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];
} t;
Then waste becomes t.waste and buf becomes t.buf. The rest of the code
then stays unchanged, in particular not having to add or subtract
AST_FRIENDLY_OFFSET in various places.
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Is BJ Weschke on vacation? I haven't seen him here for a while.
The reason I ask is because a bug of mine #6731 is assigned to
him, and I see other things happening in meetme in the meantime,
which means I have to keep my patch up to date!
Just curious...
Cheers
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In article [EMAIL PROTECTED],
Kevin P. Fleming [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
I thought non-static functions that were available as part of the C API
for use in other modules should all be named ast_something()...
It _is_ a static function, in res_features only, IIRC
would only take effect if its
value were less than the length of the second parameter.
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.
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/pipermail/asterisk-users/2006-February/145787.html
I believe Kris has made available his scripts to automate the process.
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with 2.6 kernels. They do run under 2.4 kernels, but you still have
to be careful of timing issues with some drivers.
(No, I'm not an expert - I'm still fighting with this stuff myself!)
Cheers
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appreciate it! Thanks.
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interface.
I thought I had seen some talk of such a feature in the past, but can't
find any reference to it now.
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In article [EMAIL PROTECTED],
Jeffrey C. Ollie [EMAIL PROTECTED] wrote:
On Wed, 2006-02-15 at 10:44 +, Tony Mountifield wrote:
Has anyone done any work on enhancing the MeetMe keypad menu to allow
the initiation of an outgoing call which will be connected to the
conference? e.g. *5
are getting discarded after the point where tcpdump sees them.
I'll continue investigating, but wanted to see if anyone else might have
any ideas.
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Hi Paul,
Thanks for your reply:
Paul Cadach [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
[skipped]
When I start up zaptel, with wcfxo, ztd-eth, ztdynamic and zaptel
loaded, my monitoring machine initially sees one TDMoE packet per
millisecond, as expected. After between 75 and 90
on this question, i apologize... it just seem strange.
Hope this helps!
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In article [EMAIL PROTECTED],
John covici [EMAIL PROTECTED] wrote:
What about using trunk -- will that get me 1.2.2 or netsec of both?
Using trunk will get you all the bleeding-edge stuff that will never
make it into 1.2, but is being got ready for 1.4.
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the v1-2 branch in CVS,
or else you update to 1.2.1 when it comes out. V1.2.1 is, by definition,
V1.2.0 plus subsequent bug fixes.
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.
It would be nice to get them consistent, or to have both forms accepted
by both interfaces.
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In article [EMAIL PROTECTED],
Kevin Bockman [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Chih-Wei Huang [EMAIL PROTECTED] wrote:
Since the bug is closed and seems I can't reopen it,
I post my test result here.
The version under test is 1.2.0-rc1
2.
There could have in the past been systems where long is bigger than int,
but never the opposite.
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;
+ else if (res -32767)
+ *input = -32767;
+ else
+ *input = (short) res;
+}
+
+static inline void ast_slinear_saturated_divide(short *input, short value)
+{
+ *input /= value;
+}
extern int test_for_thread_safety(void);
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there are other applications that also don't.
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with Q.SIG, I can't tell whether this support is
complete or partial.
So, can anyone tell me how complete Q.SIG support is in Asterisk CVS Head?
In case it's relevant, it will be connected to an Ericsson BP250 which
has a Q.SIG module installed.
Thanks
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for your quick responses!
The application is just for a simple conference bridge to sit behind the
PBX for people to dial into. So I expect the advanced Q.SIG features will
not be required. I hope not, anyway.
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because
they know that's the only place where the knowlegable folks hang out.
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-dd
You can look for likely starting points with cvs log chan_zap.c.
Hope this helps!
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, ChanRelease or is hung up.
I don't have time to produce them, but thought I'd toss the idea out in
case anyone feels like picking it up.
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to obtain the benefits. :-(
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the global variable DB_RESULT.\n,
[...]
+ also set the global variable DB_RESULT to that value if it
exists.\n,
Shouldn't the above two descriptions say channel variable instead of
global variable?
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module called ztrtc, and only for 2.6.
2. Make it an update to ztdummy.c, replacing the add_timer code with
the rtc_request() code.
Any recommendations which of the two I should do?
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In article [EMAIL PROTECTED],
Luigi Rizzo [EMAIL PROTECTED] wrote:
On Mon, May 16, 2005 at 11:06:16AM +, Tony Mountifield wrote:
I've been using MeetMe via IAX with no problems on a FC1 box with the
2.4 kernel and zaprtc for timing.
Recently I've set up a FC3 box with the 2.6 kernel
In article [EMAIL PROTECTED],
Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On May 16, 2005 08:59 am, Tony Mountifield wrote:
I think what is happening is that the zaptel processing invoked by ztdummy
is not happening quite often enough due to missed jiffies. Consequently
I suspect the incoming
the procedure works.
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if
the dialplan does a Goto out of the h extension to somewhere else.
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in the
conference (assuming he is in the conf to talk as well as listen)?
It might help people to suggest approaches if you explain what problem
you are trying to solve - what you are fundamentally trying to achieve.
It may still be a -users topic, but we can't tell at the moment.
Cheers
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has a project to do this. If so, perhaps we can collaborate? Anyone
have any info on this?
Surely the IAX2 code has been implemented against some kind of spec, even
if that spec exists only at Digium. There must be design documents...
Or is it all just in Mark's head?
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something like asterisk-nozap, and one built with libpri and zaptel
available, specifying them as prereqs.
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around for a while. Open a bug report on it and
post your patch there. That way there is a better record of the updates.
Yes, that's this morning's job. :-)
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the same length as the enter
sound.
I don't know whether the same issue applies to direct Zap channels or not.
Any comments?
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In article [EMAIL PROTECTED],
Kevin P. Fleming [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
I am suspecting that the problem is something to do with the conf_play()
of the enter and leave sounds. My guess is that by writing that raw data
into the pseudo device fd, it causes a backlog
In article [EMAIL PROTECTED],
Tony Mountifield [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
Kevin P. Fleming [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
I am suspecting that the problem is something to do with the conf_play()
of the enter and leave sounds. My guess
Does anyone know if there has been any work done on enhancing format_wav
to support MS-ADPCM format (fmt codec type 2)?
If not, I may consider having a go myself, but I don't want to re-invent
the wheel unnecessarily.
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.
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there was a better overall approach.
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like I would
be duplicating a lot of the functionality of app_dial. Unless there
is a way I can somehow pipeline into the existing app_dial. There are
large parts of the Asterisk code that I have yet to become familiar with!
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INVOLVED is thereby disqualified from making constructive criticism.
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.
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those options the default by using a .cvsrc file. Mine is:
cvs -z3
update -d -P
checkout -P
diff -u
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Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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.
When compiling kernel modules under Fedora you should use gcc32 as the
compiler, as that is what the kernel is compiled with. In the zaptel
directory use the make command:
make HOSTCC=gcc32
or else change the HOSTCC line in the makefile to specify gcc32.
Cheers,
Tony
--
Tony Mountifield
Work
In article [EMAIL PROTECTED],
Steven Critchfield [EMAIL PROTECTED] wrote:
On Thu, 2004-02-26 at 15:35, Tony Mountifield wrote:
But I don't see any way to use MeetMeCount in this context, which is
why I'm thinking of enhancing the code. Similarly for conditional gotos
based on time: I
be using
meetme.conf to control which rooms are available at any one time.
Cheers,
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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