similar to: Dialing PIN from console

Helo WipeOut, I have found a solution for sending dtmf after dial. I use spooling. Take a look at the sample.call file inside asterisk dir. You need to edit this file and dump it in /var/spool/asterisk/outgoing. Asterisk will precess this file automaticlly I create the sample.call do something like this: Channel: OH323/4324324324 #dial the access way MaxRetries: 3 RetryTime: 60 WaitTime: 30

I am trying implement two-stage dialing. Scenario is following: 1. * Dials SIP agent 2. SIP agent answer the phone and provide dial tone 3. * Sends DTMF string 4. "Bridge" channel with calling party I thought that something like: exten => _2XX,2,Dial_but_not_connect_(SIP/BYEXTENSION,10) exten => _2XX,3,Wait,1 exten => _2XX,4,SendDTMF($DTMF_DIGITS) Should do it. Thank

On Tuesday 23 May 2017 at 19:20:25, Tech Support wrote: > All; > > What I did was add a line in the dialplan that used the SendDTMF() > application and that worked great. The problem that I?ve run into now is > that dialing the extension screwed up the answering machine detection. The > sample context looks something like this. > > [play-audiomsg] > exten =>

Hi list, I am facing the following problem, i need to make the following scenerio work exten => _900.,1,Dial(Zap/g1/xxxxxxxxxx,25,r) exten => _900.,2,Wait(3) exten => _900.,3,SendDTMF(${EXTEN:1}) I am using PRI-ISDN with T400P card. Searching through archives, i found that we can add some 'w' s to the dial string. I tried that using on both PRI and Analogue channels but it

Hi All, I have a slight problem when trying to dial out. When I dial any number out I get only a dial tone and the number is not dialed I have to then dial it manually. I have tried my extension.conf with my pstn box and it works fine but for some reason it won't with the isdn card. I'm using the fritz pci card. Has any one else had this problem in the past??. I have also tried to set

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

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

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

There isn't much documentation on adsi, but I called NETXUSA (the vendor of my 480e) and they helped me along. My experience: 1. I really had no experience with ADSI so I had (probably still have) some misconceptions on how the configuration is loaded onto the phone. 2. I set the following in my /etc/asterisk/asterisk.adsi (most of this is the stock asterisk.adsi script): ;

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

Hi, I've solved the flash transfer problem changing the flash time in the zapata.conf file, I've set: flash = 200 (the defualt was 750 ms) in the extensions.conf the code is for example: exten => 42,1,Flash() exten => 42,2,SendDTMF(42,250) exten => 42,3,Hangup() now the transfer with flash works correctly. About the question whether my PBX expects a hook flash for

I have an X100P connected to an extension of a Panasonic PBX. When a call from the PSTN comes in, it is routed directly to the extension where the x100p is . I want * to answer the call, play a message and then transfer the call to another extension via the Zap channel where the call was received (I need to flash the zap channel) . If this extension doesn't answer I want then to dial an IAX

for the particular configuration of software/hardware that connects to my asterisk pstn gateway I need to do something like the following : [...] exten => _X,3,Dial(CAPI/02xxx.b${EXTEN},60,M(senddtmf)) [...] [macro-senddtmf] exten => s,1,SendDTMF(*) but the DTMF must be sended to the caller channel, and not the called : SIP -> * -> ISDN SIP calls some ISDN number, when ISDN picks

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

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

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

At 02:36 PM 9/14/2020, you wrote: >On 2020-09-14 16:52, Robert Heller wrote: >>At Mon, 14 Sep 2020 13:14:44 -0700 CentOS mailing list >><centos at centos.org> wrote: >> >>>Folks >>>I've encountered situations where I want to reuse a hard-drive. I do > >If it is a Seagate, don't bother. They have the highest failure >rate in the