How often were you envisioning having the job call home? I was
thinking at startup and core reload and if that's the case why bother
with an external process and the work that goes with it? I know
someone mentioned they don't want asterisk crashing because of a flaw
in the beacon but you
On Tue, May 12, 2015 at 1:47 PM, Tzafrir Cohen tzafrir.co...@xorcom.com
wrote:
On Tue, May 12, 2015 at 11:15:40AM -0600, George Joseph wrote:
The only volatile stats to be reported were active/total calls and I'd be
willing to give that up to keep the process simple. Even if we did keep
On Fri, May 8, 2015 at 10:37 PM, Matthew Jordan mjor...@digium.com wrote:
1) Not all of the data you may want to retrieve may be readily
available. Granted, right now, I don't think that's the case, but
there's certainly more flexibility if you have access to Asterisk's C
APIs.
If the time
On Thu, May 7, 2015 at 10:35 PM, Matthew Jordan mjor...@digium.com wrote:
Hey everyone -
At the past several AstriDevCon events, we've had an open discussion
about adding a module to Asterisk that would gather anonymous usage
statistics. Said module would be used to help the Asterisk
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Ship it!
Ship It!
- Moises Silva
On Feb. 18, 2015, 10:32
20, 2014 on Asterisk 11 (see
ASTERISK-21930 comment)
Thanks,
Moises Silva
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On March 3, 2014, 1:19 a.m., Moises Silva wrote:
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the ~180 you
have here.
Moises Silva wrote:
Fixed ... but, doesn't everybody has huge screens now days? ... I agree
could be handy if I am reading code on my phone though :)
wdoekes wrote:
Thanks :)
Some people use those screens for multiple windows at once
://reviewboard.asterisk.org/r/3248/#review11041
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On March 3, 2014, 1:19 a.m., Moises Silva wrote:
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- Moises
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On Feb. 22, 2014, 8:03 p.m., Moises Silva wrote
)
Thanks,
Moises Silva
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On March 3, 2014, 1:19 a.m., Moises Silva wrote:
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You have a lot of examples in the codecs/ directory of Asterisk.
Basically you just need to implement the codec interface specified in
the struct ast_translator. Red the comments in
include/asterisk/translate.h
Regards,
Moisés Silva
On Feb 6, 2008 7:01 AM, [EMAIL PROTECTED] wrote:
hello,
i
That may be true, but AGI is still a local-channel interface. Muddying the
water by including other channels is overkill. Using a separate interface for
global data is the logical outcome. That is, you get information that is
germane only to your specific connection on the generated
Adhearsion, but sounds like a cool framework.
Moises Silva
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-Issue 5841, Bridge two channels via a Dialplan app or an AMI event -
Russell to investigate
Russell: I have been trying to get this into trunk for some time, let
me know if you need any help with this one.
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I have just uploaded a new patch to implement the Bridge application
and manager action, can somebody please take a look at it and
hopefully make some tests? This bug/feature request has been there for
ages, im willing to have it in trunk.
http://bugs.digium.com/view.php?id=5841
Kind Regards
/06, Tony Mountifield [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
Moises Silva [EMAIL PROTECTED] wrote:
Tony, thanks for the suggestion. Yes, I remembered an issue with ILBC,
but the phone on the other side is using ULAW as well. I tried to
avoid any transcoding to see if that helped
On 11/6/06, Tilghman Lesher [EMAIL PROTECTED] wrote:
One could use the ast_smoother interface. It was designed specifically
to take frames of variant sizes and produce frames of a single expected
size, precisely what you want to do.
The basic interface is:
Initialize with ast_smoother_new(),
On 11/6/06, Dan Austin [EMAIL PROTECTED] wrote:
In the scenario that Moises described, it sounds
like he might managed the server that the SPA is registered to. If
he had the ability to force the framing between the two servers to
20ms, it would have helped. It would not help if an endpoint
second ), this is needed because the
mix_slinear_frames function in app_conference requires that maximum
number of samples. Any ideas, again to make it work with iLBC or any
other frame size dependant codec?
Thanks.
On 11/6/06, Moises Silva [EMAIL PROTECTED] wrote:
On 11/6/06, Dan Austin [EMAIL
executing applications on the
PBX, so, is not a thread for each application, but a thread for each
channel in the PBX. Note that some applications, like Dial(), create
other channels, but in the same thread as the application executor.
Kind Regards
Moises Silva
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/archives/2005/06/new_download_--_3.html
Just need something else to read from? But, could you show me some use
for it, im missing that :(
Thanks
Moises Silva
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Thanks for working on this. I had been trying to get this feature into
Asterisk over a year ago and gave up because none of the core
developers were interested in adding it.
So you are user heath1444 in Mantis?
Maybe if we get enough support behind this it will finally make it into
Thanks for the response Russell. I have opened this bug:
http://bugs.digium.com/view.php?id=7855
With a patch for the stable branch (1.2.11)
Tomorrow I will check the solution you suggest for Trunk, when done, i
guess i need to open a new bug for that right?
Regards
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Youre right, chan_sip.c is the right place. But AFAIK Asterisk does
not support silence suppression. If you remove that asterisk may stop
sending sound, because it seems it uses the incoming voice frames
(including silence frames) as timer to send its own sound frames.
Plase correct me if im
hum, i guess this does the trick:
http://www.voip-info.org/wiki-Asterisk+cmd+SetAccount
Is deprecated, but there you find other way to do it.
regards.
On 1/4/06, Cristian Draghici [EMAIL PROTECTED] wrote:
Hi list
Is there a way to alter the accountcode CDR field from AGI?
I didn't find
Or you can use MAGI patch, to execute agi stuff from the manager :)
On 1/4/06, Alexander Lopez [EMAIL PROTECTED] wrote:
You can execute an AGI script BEFORE you answer the call, it should only
delay the call a ms or so.
In this AGI you can either save the values to a DB or send them
Hi Franco. I have a PHP script that connects successfully and receives
the responses correctly. I think that it would help you to read the
manager.c file in the Asterisk source code. Also, its difficult to
know why you are not getting the whole response from asterisk, since
the code you show us
Hi Pere.
I have been using the manager for a few days, but i have been doing
some research into the source code. Its interesting. The function
that send events is
int manager_event(int category, char *event, char *fmt, ...)
located in manager.c
i use grep command to find the events that are
Hi. Not sure about this. But i have been using some spanish sounds
that i have downloaded some weeks ago (i dont remember where), and
does not exists the sound 'vm-youhaveno.gsm', instead has the sounds
'vm-youhave.gsm' and 'vm-no.gsm' just as the english version does. I
realized of this when
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