search for: senddtmf

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 happe...

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 C...

...d 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 having a second priority with SendDTMF. 2) How can I silence the first seconds of this call (so I do not have to listen to their stuff)? or play some music on hold? 3) Another diferent problem I would like to solve is how to park and incoming call without answering, so I can call Asterisk from my cellphone, dial the 700 extension a...

...e 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 extension.conf are as follows: [test] exten => _0XXXXXXXX,1,Dial(H323/${EXTEN:0}) exten => _0XXXXXXXX,2,SendDTMF(PIN_NUMBER_HERE) .... I saw someone post the similiar question but with no reply. Does anybody has any idea? cheers Foong -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030805/81ddeb98/attachment.htm

...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) exten=>t,1,NoCDR() exten=>t,2,Hangup() exten=>_009[13456789].,1,Dial(SIP/operador/${EXTEN},60,tr) exten=>_009[2].,1,Dial(SIP/operador/${EXTEN},60,tr) exten=>_00[12345678].,1,Dial(SIP/operador/${EXTEN},60,tr) exten=>_6[0123456789].,1,Dial(SIP/operador/${EXTEN},60,tr) exten=>_...

...VR. 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 execute the following code sequence: exten => _7XXX,1,Flash() exten => _7XXX,2,SendDTMF(${EXTEN}) exten => _7XXX,3,HangUp() where _7XXX is a phone within the Bosch system. This frees the zap channel and the caller will communicate with the transferred destination directly through the Bosch system without passing through Asterisk. However, if I setup a queue and the caller en...

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,...

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 _________________________________________________________________ Add photos to your e-mail with MSN 8. Get 2 month...

...Allof the cases I saw the `w` token discussed was at the beginning of the dialed digits. I tried putting a `w` at before the ${EXTEN} and it did appear to wait a little before dialing the digits. Maybe it doesn't work inside the dial string? I also tried the recommendation to use the Flash, SendDTMF, Wait, SendDTMF, but that didn't work at all. I need to implement this functionality in the dialplan. Does anyone have any suggestions? Thanks for your help! -- ******** DataCrest, Inc. -- Technically Superior ****************** Walker Haddock http://www.datacrest....

...all and I can dial an extension (for example 42 that is on the traditional PBX). In the asterisk dialplan I've set to transfer the call using Flash() like in this example: exten => 42,1,Flash() exten => 42,2,Background(silence/1) wait 1 second for the traditional PBX exten => 42,3,SendDTMF(42,250) exten => 42,4,Background(silence/1) wait 1 second for the traditional PBX exten => 42,5,Hangup() When I dial the extension 42, the phone 42 on the traditional PBX rings but when I answer there isn't communication with the call from the TELCO line and after a few seconds the l...

...latively easy. I have successfully built a menu structure, but cannot get Asterisk to transfer calls back to the legacy PBX (Fujitsu F9600). In essence, all I require Asterisk to do is: 1) read the extension digits entered by a caller; 2) flash the line [Flash()]; 3) dial the extension using DTMF [SendDTMF(${EXTEN})]; and, 4) hang up [Hangup()]. Unfortunately, I've not been able to make this work and was hoping someone might tell me where I'm going wrong. The problem appears to be in the "flash" portion of the above procedure. Asterisk Server Setup -------- ------ ----- - Ubuntu...

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

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 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 to happen after the hangup() - which is needed to close down the voicemail() call. I've tried: exten => 200,1,voicemail(200) exten => 200,2,Hangup exten => 200,3,Wait(2) exten => 200,4,Answer exten => 200,5,SendDTMF(2...

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

...e X100P. This is in the USA. Call the asterisk box. Line is answered. It can be heard with the Asterisk attempts to Flash, but it does not. The section from extensions.conf is as follows: exten => s,1,Answer exten => s,2,Wait(2) exten => s,3,Flash exten => s,4,Wait(1) exten => s,5,SendDTMF(8889995555) exten => s,6,Wait(1) exten => s,7,Flash exten => s,8,Wait(1) exten => s,9,Hangup The call as answered by Asterisk in debug mode is as follows: Asterisk Ready. *CLI> -- Starting simple switch on 'Zap/1-1' -- Executing Answer("Zap/1-1", "...

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

...sk -> 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 used for the blind transfer (hookflash) --- [from-sip] exten => _NXXX,1,Flash() exten => _NXXX,n,SendDTMF(${EXTEN}) exten => _NXXX,n,Hangup() [from-zap] exten => s,1,Dial(SIP/sv0071iv) exten => s,n,Dial(SIP/sv0072iv) exten => s,n,Goto(AA,s,1) [AA] exten => s,1,Wait(.5) exten => s,n,Background(vm-whichbox) exten => s,n,WaitExten exten => _5XXX,1,Playback(transfer) exten =>...

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 Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site...

...k 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 the following dialplan entries, that I can dial into: exten => s,1,Answer exten => s,2,Wait(2) exten => s,3,SendDTMF(1234567890) exten => s,4,Wait(2) exten => s,5,Hangup What happens is that whatever string I give to SendDTMF, I only hear the first DTMF digit. The remaining digits don't get sent. I recently updated from an April CVS-STABLE to the July 4 version, but I couldn't see any relevant dif...