similar to: SendDTMF into a conference room

Hi, I''ve got a question about the find() method in RoR This is my code: actions=Action.find(:all, :conditions => "ActivityID=''actid''") This should return a list, array, hash, ... ? of Actions. What type of data does find() return? And how do I run through it? I would like to show Action.ActionCode for every action in actions. I tried: for action in

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

Hi everyone: The following code dials our prefix, sends a beep, and sends a DTMF "c" tone, then dials the phone number. I need to send the DTMF only if the phone is answered. [voip] exten=>i,1,NoCDR() exten=>i,2,Hangup() exten=>s,1,Wait(2) exten=>s,2,Background(beep||) exten=>s,3,DigitTimeout(6) exten=>s,4,ResponseTimeout(10) exten=>s,5,SendDTMF(c)

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

On 14/06/16 18:36, Juan Ignacio wrote: > Traducir del: Inglés > Ok, I have correctly configured the option to use RFC2307 in the > Actide Directory Domain Controller and the option on the member server. > > But anyway still showing bad UIDs. > > Here we show how both configurations are actualemente. > > Within RSAT I can place the UID attribute for example uanaco. >

On 14/06/16 17:32, Juan Ignacio wrote: > Rowland, a question. > > > "is to copy idmap.ldap from the first DC to all others and then keep > them in sync, the other is to use RFC2307 attributes." > > I can do the same with my member server? Maybe it works, or not for > beign a member server. > > Maybe i can change my Member Server to a Domain Controller and

I have a dialplan that requires that we use * to send the long distance access code to AT&T. I have found in the list that the `w` command can be used to inject a pause, I have tried the following: exten => _91NXXXXXXXXX,1,Dial(ZAP/g1/${EXTEN}www5555555,70) There `5555555` is the ld access code. I tried various quantities of `w`s but I never got * to dial the ld access code. Allof the

I've tried to transfer a call using the Flash command, but with my configuration it doesn't work. I have a traditional PBX connected with a zap channel to Asterisk that acts like an IVR: TELCO line --> traditional PBX (FXS) --> (FXO) Asterisk >From the TELCO line I can make a call to the traditional PBX and reach Asterisk, the IVR system on Asterisk answers the call and I can

Greetings. I am attempting to configure a system based on Asterisk 1.2.3 to be used as a backup should our aging voice mail/auto attendant system fail, which seems increasingly likely given its advanced years. The first part of this task is getting the auto attendant feature to work correctly, which I would have figured to be relatively easy. I have successfully built a menu structure, but cannot

Hello, I'm looking for a similar feature... Dial a number via ZAP/g1 after the line gets answered wait 10 seconds send DTMF Regards, Marc -- Network Manager Marc Storck LuxAdmin.Org mstorck@luxadmin.org Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352

This 2 line code is doing what I wanted. exten => 200,1,voicemail(200) exten => 200,2,Hangup What I've been told is that they want the 20 year old phone system to light up the message bulb. (yea, a filament bulb, not an LED) To do this you pick up on the line that goes into Asterisk and do a: exten => 200,1,SendDTMF(200w#86) But I don't know the path to take to get that

Problem: Flash on X100P does not flash. Phone line has Call Transfer. With this line plugged into a regular phone, it can receive a phone call. Then, depress the hook momentarily, release. Dialtone is now available. Dial a different number. Call is answered. Hook Flash again, now in a three way call. Hang up. The other two parties are still in communication. Now, plug same line into the X100P.

I need to be able to dial a long distance provider that uses an access number and an auth code. I would like to be able to program this so that the user can dial 8 and then the long distance number, asterisk will hopefully do everything in the middle. The sequence to accessing the provider is on my traditional phone speed dial as: * Dial local access number * Wait 5 seconds * Dial the auth

List, Having a little trouble with the following. Let me prefix by saying I have blind transfers working from the following setup. Inbound call [from-zap] (SIP/sv0071iv) answers. Zaptel -> Asterisk -> SIP extension SIP extension then blind transfers [from-sip] --- SIP extension -> Asterisk -> Zaptel During this whole process, the original channel off the trunk (lineside T1) is