Displaying 20 results from an estimated 166 matches for "canal".
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anal
2006 Mar 23
0
sqlSave
Dear Tobias,
I finally succeeded in exporting my dataframes from R into Access with
this script:
library(RODBC)
canal <- odbcConnectAccess("D:/Floristique.mdb")
sqlSave(channel=canal, dat=flore, tablename="Floristique", rownames=F,
safer=F, fast=F,
varTypes=c(dates="Date"))
odbcClose(canal)
My problem in exporting my dataframe "flore" seems to be closely linked...
2006 Mar 18
1
The R "fork"
Hello, I would like to call a function that can take infinite time to
be executed in some circumstances (which can not be easily detected).
So, I would like that once the function is being executed for more
than two seconds it is stopped. I have found documentation for timers
but i did not found how to "kill" a function. Moreover, I would like
not to change the function code (it should
2004 Apr 01
15
ANNOUNCE: Flash Operator Panel
http://sip.house.com.ar/operator
Its a server/client combo that displays the status of your Asterisk PBX
in a web browser in real time.
You can also perform some actions. Hang-up channels and Transfers via
drag and drop.
The difference with other similar tools is that it displays status in
real time (no refreshing necessary), and its graphically appealing.
It's a work in progress... so
2008 Jan 09
0
[asterisk-dev] MixMonitor doesn't work right with SIP and Zap/Flash transfers
...sterisk box, so I use the
> MixMonitor() application like this is my extensions.conf:
>
> exten => _0X.,1,Answer()
> exten => _0X.,n,MixMonitor(${CALLERID(num)}-${STRFTIME($
> {EPOCH},America/Sao_Paulo,%Y-%m-%d-%H-%M-%S)}-${EXTEN}.wav)
> exten => _0X.,n,Dial(IAX2/pabx-canall/${EXTEN},60,tT)
>
> exten => _2XX,1,Answer() exten => _2XX,n,MixMonitor(${CALLERID(num)}-$
> {STRFTIME(${EPOCH},America/Sao_Paulo,%Y-%m-%d-%H-%M-%S)}-${EXTEN}.wav)
> exten => _2XX,n,Dial(SIP/${EXTEN},60,tT)
>
> The scenario is as following:
>
> 1) 201 asks op...
2004 Dec 07
3
Question about e1/digium
Hi all I am beginning in asterisk and am making tests with an ata-186.
For the time being the tests are going well, however have a doubt.
I am thinking about using a canal e1 with plate digium.
Assuming that the company of telecommunications supplies e1 with 30 canals
and numeration to me 4000-0001 4000-0029. she is possible to configure
asterisk
in way that somebody of is dials 4000-0025, to direct for a telephone sip ?
Thanks for attencion
Sergio Faulhaber
serfa...
2011 Jun 09
0
IRC canal
Hy all of you,
I try to access to the IRC canal without success. Where should I
subscribe to be allowed to chat?
Regards
Florent THOMAS
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2009 Jun 02
9
XML::LibXML::Reader
is there anybody who knows how we can get child values using libxml
(Reader class)
def xml_import_to_brands
require''xml''
reader = XML::Reader.file(''c:/brands.xml'')
.
.
.
2008 May 05
3
TDM410P driver?
Quick and dirty question: for the TDM410P I must use the wctdm24xxp driver?
Att
Vin?cius Fontes
Desenvolvimento
Canall Tecnologia em Comunica??es Ltda.
2004 Apr 02
1
convert excell file to text with RODBC package
...55 8,3 35 79,6 86,4
15/12/2003 12:15:04 217 2766 8,3 35 79,6 86,4
15/12/2003 12:15:05 217,1 2766 8,3 35,1 79,6 86,4
15/12/2003 12:15:06 217,1 2758 8,3 35,1 79,6 86,4
15/12/2003 12:15:07 217,1 2768 8,3 35,1 79,6 86,4
My code (i use the R 1.7.1 for windows xp) is the following:
library(RODBC)
> canal<-odbcConnectExcel("c:/prueba4.xls")
> tablas<-sqlTables(canal)
> tablas
TABLE_CAT TABLE_SCHEM TABLE_NAME TABLE_TYPE REMARKS
1 c:\\prueba4 <NA> Hoja1$ SYSTEM TABLE <NA>
2 c:\\prueba4 <NA> Hoja2$ SYSTEM TABLE <NA>
3 c:\\p...
2009 Oct 30
7
Voicemail file
Hi all,
When somebody leaves a message in the voicemailbox, is there a way to know the file name of it?
I need to return the voicemail file name in the deadagi command.
Thanks,
Anahi
_________________________________________________________________
Convierte las fotos que m?s te gustan en tu nuevo fondo de escritorio para el ordenador. Es f?cil y adem?s gratis
2004 Apr 02
1
ANNOUNCE: Flash Operator Panel - Extensions fixed
...ake care of that, my asterisk universe is very limited,
I
>did not think about other naming conventions and uses for the different
>types of channels.
I was able to get the extension/channel problem fixed. In version .03,
change the following lines, starting at line 329:
Old:
while ( ( $canal, $nroboton ) = each(%botones) ) {
if ( $nroboton eq $destino ) {
$canal_destino = $canal;
log_debug( "Si!!", 4 );
}
}
$canal_destino =~ s/.*\/(\d*)/$1/g;
N...
2002 Sep 25
0
solaris/2.2.2 authentication against NT domain (ie. r oot canal)
why do you have smbpasswd file, when you are in domain mode?
-----Original Message-----
From: AMueller@YesVirginia.org [mailto:AMueller@YesVirginia.org]
Sent: Wednesday, September 25, 2002 3:07 PM
To: samba@lists.samba.org
Subject: [Samba] solaris/2.2.2 authentication against NT domain (ie.
root canal)
Greetings,
This is my first post (hopefully not too newbie for this group). Thank you
in advance to anyone who feels compelled to help. I'm trying to get a copy
of Samba 2.2.2 running on Solaris 9 to authenticate against an NT domain. I
have successfully done this on a Redhat 7.2 box wh...
2008 Apr 24
1
Full queue issues
...enters the queue and stays there. I would like the call NOT to enter the queue and the ${QUEUESTATUS} variable to be set to something different.
Am I missing something or it's just not possible? I'm using SIP phones for the agents and Asterisk 1.4.15.
Att
Vin?cius Fontes
Desenvolvimento
Canall Tecnologia em Comunica??es Ltda.
2017 Feb 14
2
pregunta
Estimados
Adjunto en excel el canal endémico de Bortman, se brindan las tasas de
cinco aaños anteriores de cualquier enfermedad endémica y no0s alerta si
nos alejamos de la media histórica (media geométrica) los intervalos de
confianza son con conversión logaritmica.
¿Es posible traducir este excel a R?
saludos
José
--
Este mensa...
2010 Mar 06
0
Audio problems ins conference zap->sip
Hello!
I have several problems in the audio one belonging to asterisk at
conferences between ZAP - SIP. I hope that you may help me.
1 Problem
When the audio establishes a call between two canals, some zap and another
sip itself one listens interrupted in one of the senses, exactly in zap sip,
with audio cuts. Words lose at random, in that connection, however in the
sense sip zap the conversation is listened to perfectly without cuts.
Sip in one of the extremes and a equipment in the othe...
2000 Aug 24
2
hosts allow/deny question
...* is not understood by Samba, I tried ALL, this didn't
work either. I'm gonna check the samba source code but if I could
get an expert answer that'd be nice =)
Any idea ?
Ben.
--
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=+=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
=- Benoit Boudeville | CANAL+ Technologies -=
-= Computer System Engineer | 34, place Raoul Dautry =-
=- mailto:bboudev@canal-plus.fr | 75516 Paris Cedex 15 -=
-= Tel: 01.71.71.55.83 | Fax: 01.71.71.55.77 =-
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=+=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
2009 Jun 10
2
add variable in xpath
hi ,
i cant succeed add variable in xpath (libxml)
brand = "037"
xpath = "//Root/Stk[@Category=''601'' and @Brand=#{brand}]"
stack = node.find(xpath)
i used #{brand} but doesnt work
2002 Sep 25
1
solaris/2.2.2 authentication against NT domain (ie. root canal)
Greetings,
This is my first post (hopefully not too newbie for this group). Thank you
in advance to anyone who feels compelled to help. I'm trying to get a copy
of Samba 2.2.2 running on Solaris 9 to authenticate against an NT domain. I
have successfully done this on a Redhat 7.2 box which is running Samba 2.2.4
on the same domain. As best as I can tell, the configurations of these
2010 Feb 05
6
Still on spandsp/app_fax and T.38
...itting at full 9600bps. Just to rule out Asterisk's T.38 handling, I noticed that Fax For Asterisk works perfectly with T.38, also on that same box.
I'll be happy to provide you any info that could help solve this issue.
Atenciosamente,
Vin?cius Fontes
Gerente de Seguran?a da Informa??o
Canall Tecnologia em Comunica??es
Passo Fundo - RS - Brasil
+55 54 2104-7000
Information Security Manager
Canall Tecnologia em Comunica??es
Passo Fundo - RS - Brazil
+55 54 2104-7000
2010 Feb 05
8
Losing local SIP phones when internet goes down?
Hi,
I'm getting some strange behaviour on Asterisk 1.4 running on Debian
Stable (Lenny). I suspect it's something to do with my setup, rather than
a bug, but I'm struggling to see it, and would appreciate any input.
Setup: PC with two ethernet cards: eth0 goes to local network, including
two SIP phones (Aastra 9112i, wired, and Nokia E75, over WIFI); eth1 goes
to router and