similar to: flash transfer problem in asterisk integration with old PBX

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

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

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

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

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

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

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.

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 =>

I'm not sure if this is a -users or a -dev question, since the answer probably comes down to something in the code. I'm running the latest CVS-STABLE, and am subscribed to PSTN service using IAX2 via Voiptalk in the UK. I've just been alerted by a customer that the sending of DTMF from my asterisk box to a remote PSTN user doesn't work, although it used to. To test it, I have

Hi, After some testing I've found out that my client's hardware recognizes DTMF only if digits are sent 50ms apart with 50ms of tone duration. This was tested using a test device which generates DTMF. Now asterisk doesn't do it by default because digits going out from Asterisk are not being recognized. Using command sendDTMF, I can control inter-digit duration, and using

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 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 am working with the telantek.adsi file, and I was wondering how I would create a softkey for Transfer. I tried making a key definition and using SENDDTMF "#", but that didn't work. Is there another way I could do this? Also, does anybody have any ADSI scripts for use with Asterisk that they would like to share? Thank you for your time. __________________________________ Do you

Dear All I have this setup [asterisk]----[mediant2000]-------E1 Trunk----------[Avaya] When i call from avaya to asterisk i got long ringing tone then hangup but when i call from asterisk to avaya i got 4 ringback and then hangup with this error *CLI> Jun 26 01:26:08 NOTICE[5555]: chan_local.c:523 local_alloc: No such extension/context 1022 at mysip

Hi list, I would like to know of a possible way to dial a pstn number with an extension . Let the number is 56626965-234 so now i wanna dial 56636965 then wait for some time and dial the extension 234 to reach a particular person.I am afraid that i could not figure it out. I am trying in this way.. [outgoing] exten=>_566X.,1,wait,2 exten=>_566X.,2,Dial(${EXTEN})

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

I am trying to send a DTMF digit automatically every 15 seconds to keep a call connected to an alarm panel. I tried using the dial command L and recording a dtmf tone for the beep, but obviously that didn't work. Does anyone have a suggestion for merging the L option and the sendDTMF or the D option? Any other suggestions would be appreciated! Thanks! Paul Gentilini

I'm trying to call a conference service, wait 10 seconds, then send the passcode. I've tried ww: Dial(SIP/18005551212ww12345#@sip.com,60,r) The sip channel didn't like that. Added 'p' , still no help. I tried D: Dial(SIP/18005551212 at sip.com,60,rD(12345#) The dtmf is sent too soon. I tried inserting 'ww' but that was just sent. I tried G: exten =>