similar to: sendDTMF at pickup

Hi, I got the resource "Llamadas" nested in: - Operadores - Productos - Centros Here is part of my routes http://pastie.caboo.se/92767 I want to spec that the routes for Llamadas, I tried several approachs: - route_for(:controller => "llamadas", :action => "exitosas", :operador_id => 1).should == "/operador/1/llamadas;exitosas" -

Hello I'm having some disk full messages in several windows xp clients.The disk have a lot space free. I'm using Centos 5.5 with the samba centos official package. samba-3.0.33-3.29 Maybe a samba bug ? Any advice ? Thanks a lot for any help regards roberto This is my smb.conf #======================= Global Settings ===================================== [global] #

Que tal comunidad, una pregunta general: existe en la librería ggplot, algún comando que permita hacer simultáneamente 2 graficos dentro del mismo layout, como el operador | en la libreria lattice ? muchas gracias, un abrazo a todos, Eric.

Muchas gracias Carlos !! un abrazo, Eric. On 30-05-19 17:57, Carlos Ortega wrote: > Hola, > > Sí, en ggplot esto se resuelve con: > > * facet_grid() > * facet_wrap() > > Saludos, > Carlos Ortega > www.qualityexcellence.es <http://www.qualityexcellence.es> > > El jue., 30 may. 2019 a las 23:30, neo (<ericconchamunoz en gmail.com >

Hi: Astcc is working fine, except for one thing. It doesn't give the called party enough time to answer the phone. If nobody picks up in two rings, astcc reports back no answer and hangs-up. The only instant NOANSWER "value" was mentioned in astcc.agi script is: elsif ($res eq "NOANSWER") { $res = &mystreamfile("astcc-noanswer");

Can any body give me an example how to configure h323 in Asterisk. Which files do I need to configure? just extensions.conf and h323.conf ? Thanks, Patricio _________________________________________________________________ Descubre la descarga digital con MSN Music. M?s de un mill?n de canciones. http://music.msn.es/

Buenas. Soy un chaval novato en el uso de R. Ando haciendo un curso online, y he llegado a una parte en la que me he atascado. La instrucción es la siguiente: crea una variable llamada | 'mi_vector' que contenga un vector con los números enteros del 11 al 30. Recuerda | que puedes usar el operador secuencia ':' A lo cual yo he introducido el siguiente comando:

Hola a todos! Soy un novato con R y, como suele pasar, tengo algunas dudas que me gustaría compartir con vosotros. Antes de nada decir que llevo muchas horas buscando información y peleándome con los manuales, tanto en inglés como en español, pero no sé como resolver mi problema que es el siguiente: ¿Cómo puedo hacer para crear un programa con el que, al darle una función y un punto, me diga que

Hi to all, I have a strange behavior in my asterisk server. I have a queue for 5 agents, the calls enter the queue an go to the agents normally, but if I need to transfer or dial directly to an agent extension that is already in a call, the pbx hung up the actual call (not the transferred call). This is what I see in the log. Called 103 -- Agent/103 is ringing --

Cordial saludo, Necesito de su ayuda dado que, cuando estoy trabajando en Rstudio no me están funcionando los atajos *CTRL + 1, CTRL + 2*, además cuando utilizo el atajo* CTRL + SHIFT + M* el operador que obtengo es: |>. Atajos como *CTRL + ENTER, ALT + -, ALT + SHIFT + K* funcionan perfectamente, por ende, no entiendo qué está pasando ¿Alguien sabe que puede estar ocurriendo? ¿Por qué puede

Hola a tod en s, tengo una duda que se relaciona con alternativas al uso de variables globales. En principio, si se quiere usar un generador de v.a con la librería Runuran sólo se permite definir las funciones de densidad (o el núcleo de las mismas) con funciones con un único argumento en (x). Sin embargo, necesito pasar a las funciones más argumentos que van cambiando en las iteraciones de

Hi, everybody: As part of a paging macro I'm using SendDTMF to send digits to the called party. The section looks like this: exten => s,1,Wait(0.5) exten => s,n,SendDTMF(9531290) exten => s,n,Wait(1.0) exten => s,n,Set(MACRO_RESULT=CONTINUE) To test I direct the call to a live extension just to hear what's happening -- what actually happens is that only the 9 is sent, and

I have been trying to figure a way to SendDTMF into a MeetMe room using the Manager API. I can't redirect everyone into another context and then bring them back because that would mess up my logic. I am trying to use local channels and the originate Action to accomplish this. Exten: 3441115 Priority: 1 ActionID: actid-00000001 Context: senddtmftones Action: Originate Channel:

Hi all! I would like to solve some problems: I have a sip provider that lets me make pstn calls after listening some stuff and entering a pin number: 1) How can I make Asterisk enter the pin number? Then wait 1 second and enter the phone number? I have in extensions.conf: exten => 6*,1,Dial,SIP/2002@myprovider,60,tr I have tried with w (like with ZAP channels) but it does not work, nor

Hi! I am trying to setup a simple queue in Asterisk and I'm having a small problem. Our callers come in through a Bosch PBX and are immediately transferred to an Asterisk menu/IVR. If they select the option to call a SIP phone directly (eg. entering the operator's SIP extension) then the callee/operator can transfer the call to a phone within the Bosch system. What Asterisk does is

I'm not sure why but my call is being ended when I SendDTMF(*). I'm using agi to originate a call and set the context,extension,priority to test,1,1 respectively. I've got the following in my extensions.conf: [test] exten => 1,1,Answer(); same =>n,Wait(5); same =>n,Verbose(1, Sending *); same =>n,SendDTMF(*,500); same =>n,Verbose(1, Sent *); same

Hello all, I am trying to use asterisk to call a local access gateway by dialing a fix number, after getting connected, the is a IVR prompt for pin number and finally the real destination number. I manage to use asterisk to dial to the gateway but have no idea how to send the pin number and destination number. This is due to asterisk only process the next ext only if dial app has terminated. My

Hi Here is my scenario Mr.X's Asterisk Box Dials Mr Y's Asterisk Box (thru Zaptel channels)after Channel establishment Mr. X send DTMf tones to Mr Y using by using application "SendDTMF()". My question is this is there any method that Mr. Y Saves these DTMF Tones in any variable (after converting back to their Corrosponding Digits). Thanking in advance Obaid

Hi, Does anyone know of a way to implement the following: * an incoming call is connected to an internal extension (the internal channel is the target of the "dial") * Asterisk listens for DTMF generated by the internal extension (the dialed party) * when it detects DTMF, it jumps to a new context for the dialing party; I suppose the dialed party could be hung up on, or sent to

Hey, This is weird, I wonder if anyone has an explanation? If I call a SIP server and inject DTMF with a wait in it, my router will then lock up causing asterisk to lose Internet connectivity obviously, but also making it very hard to see what happens. It appears that if there are no 'w' in the DTMF string, it doesn't lock up. Anyone have any guesses on this? I called a local