similar to: SendDtmf

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

Hi, What is the correct chan_capi dial syntax?? This is what I think it is.. exten => _90.,1,Dial(CAPI/[msn here]:${EXTEN:1}) This seems to work for local numbers.. but I have an access number for cheap long distance calls.. wich gets dialed and then the number I want to call is sent as DTMF after a few waits (w).. On my X100P I used the following.. exten => _9001.,1,Dial(Zap/1/[access

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

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)

Hello Conference again. Meetme can now limit number of users in a room. Can it also limit how long a conference session? Someone ask the same question (from achive) but doesn't have a solid answer. Foong -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030821/c1ca1383/attachment.htm

Hello, Does asterisk supports VXML? Couldn't find much resource on that on google and wiki. Thanks Foong

Hi, Do the g729 codec licenses for Asterisk work on a SIP environment (only SIP UAs running g729 + Asterisk)? I would like to buy a couple for a SIP test lab but I have not found any documentation on wether it works for SIP UAs or not. The Digium page only mentions: "The G.729 codec works with all Digium cards." Can somebody tell me please? Thanks, Ricardo Villa

Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that

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 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

hi, Can anybody pls tell me, how to increase the time gap between 2 digits when you transfer a call. ie, the operator answers the call, and presses hash key to transfer, and then enters the extension number, some times, it timeouts too quickly before the operator enters the whole extension number (may be bcos the operator is slow). I tried the following, but it doesn't seems to be helping

I'm fitting data from a mixture experiment, and I'd like to present the results in a ternary graph with contours. I found this code by Walmes Zeviani http://n4.nabble.com/Triangular-filled-contour-plot-td1557386.html which is just what I want--except I would like the axis titles and labels to be proportionately larger than the ternary graph itself, for legibility in publication. When I

Hello, I am experiencing weird problem in Asterisk 16.2, possibly a bug. Same thing works fine in Asterisk 11. Here is the situation: I have 2 extensions on 2 phones. 4 extensions in total. phone 1: 8882 8382 phone 2: 8884 8384 And I have 2 SIP trunks for outgoing calls. I want to call via SIP1 when called via 8882 or 8884, and SIP2 when called via 8382 or 8384. And one last detail. SIP1

Hi all, I am working on a filled contour plot which shows a triangular matrix data set (as shown below). Is there a possibilty to draw a triangular filled contour in a equilateral triangle (like a ternary plot)? Thanks in advance Johannes http://n4.nabble.com/file/n1557386/Bild3.png -- View this message in context: http://n4.nabble.com/Triangular-filled-contour-plot-tp1557386p1557386.html

Hello all, I have a cisco AS5300 which is register with a gatekeeper and a Asterisk server also register with the gatekeeper. PSTN ---->AS5300 ---->Gatekeeper ---->Asterisk I set up a conference room on the Asterisk sever (Room No 1234). I try to call from PSTN to AS5300, The AS5300 will call the Asterisk server through the gatekeeper. I manage to get to the start of the conference