similar to: Odd router behavior when using 'w' in SendDTMF

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

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, 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

I'm running Asterisk 1.8 from Debian. I have some analog phones connected via a TDM400P. I'm testing them with these simple extensions: exten => 600,1,Answer() same => n,Festival(This is an echo test) same => n,Festival(Hang up or press pound when you are done) same => n,Echo() same => n,Festival(Good-bye) same => n,Hangup() exten

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)

I'm hoping people can suggest some ideas for debugging a problem that I'm having with DTMF. Unlike most of the DTMF problems reported here, it has nothing to do with Asterisk interpreting DTMF. My problem is with the synthesis of DTMF tones on outbound calls on a PRI connected to a TE412P card. I'm running * 1.4.10.1 with Zaptel 1.4.4. It is important to note that these problems

Hi I'm trying to implement a Callback mechanism whereby I generate a Call file and connect an arbitrary extension with my cellphone (via a SIP Channel). If I create a .Call file that connects the channel "SIP/12345678@Provider.net" with a local extension/context I get some weird issues with DTMF tones. I've set dtmf=2833 and the codec in use is G711a. For example - I create

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

To insert long pause during dialing and submitting multiple DTMF tones, is there better solution then below: exten => _51,1,Dial(SIP/18778794590 at pstn-5665,300,D(wwwwwwwwwww1www),D(wwwwwwww005893884053811#)) I think submitting multiple DTMF tones is not allow from one command line. The first part D(wwwwwwwwwww1www) worked, but not the second one D(wwwwwwww005893884053811#) I'm trying

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

Hello folks, I'm interested in detecting tones on specific frequencies with specific timing; for example, I'd like Asterisk to dial out and when the channel starts/call connects, listen for a 1200Hz tone that plays for 100ms. Is this doable with Asterisk using something already extant? After looking through documentation, mailing lists, and some of the source I had the idea that I might

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

Hello, I did this post a long time ago but never solved the problem, so i'm trying again after something like 10 months, hopefully i'll find someone that found a solution ;-) When i call an external number that sends audio before call has been answered (like some PBX of public offices do here in italy), strange things happen: I'm using chan_capi, with Early B3 active, i can listen

On 12/21/2014 04:42 AM, Patrick Beaumont wrote: > Have you enabled DTMF logging and seen the DTMF codes being recognised by > Asterisk? I had a bunch of soft phones that I had to change to using ?sip > info? for the DTMF signalling as the RFC signalling was not always being > recognised. This would cause transfers to appear as if the user had not > dialled any digits. > > >

Greetings All! The Asterisk Call Manager works great. But I have one question for anyone who has used it. I cannot get the system to send some DTMF tones down the channel once the call has been made. Below is the script I am using to make the call, and start recording the channel. I am starting to make a system the will use asterisk to become an automatic random quality monitoring system

I'm posting here in case anyone else runs into this and needs some help. I'll probably update the voip-info Wiki pages on Toshiba integration in a bit. Asterisk 1.6 makes things a bit easier than what is on that page. I'm integrating an Asterisk server with a Toshiba Strata system at my office. Right now, it is driving some VoIP phones (Cisco ATAs with analog phones plugged into

I have an analog GSM Gateway that is connected to a normal SIP ATA device. Basically what it does is this : when you call the extension nr. of the SIP ATA port, the GSM Gateway will pick up the phone and presents a (new) dial tone, and then dials whichever DTMF tones it received. The SIP ATA ia a Grandstream HT286. I would like to use the GSM Gateway to route my outbound cellular calls, how

On 12/21/2014 11:09 AM, sean darcy wrote: > On 12/21/2014 04:42 AM, Patrick Beaumont wrote: >> Have you enabled DTMF logging and seen the DTMF codes being recognised by >> Asterisk? I had a bunch of soft phones that I had to change to using ?sip >> info? for the DTMF signalling as the RFC signalling was not always being >> recognised. This would cause transfers to appear

Does Asterisk support Radius accounting?.... -----Mensaje original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] En nombre de asterisk-users-request@lists.digium.com Enviado el: Lunes, 17 de Noviembre de 2003 12:08 p.m. Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users digest, Vol 1 #1912 - 11 msgs Send Asterisk-Users mailing list

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