similar to: Placing a call from command line and passingit to an extension if connected - Is it possible?

When I read variables in AGI scripts, I see only the follwing 13 variables agi_request agi_channel agi_language agi_type agi_uniqueid agi_callerid agi_dnid agi_rdnis agi_context agi_extension agi_priority agi_enhanced agi_accountcode beside these, I found following variables documented on several sites. agi_calleridname agi_callingpres agi_callingani2 agi_callington agi_callingtns Where can I

Sorry for asking this question again but before I blow 100 dollars on a X100P I need to know this info: So does "SET EXTENSION <new extension>" allow for you to set which extension the rest of the call will occur over? So if a call comes into the switch and I could make the AGI script check the DID or DNIS which is really in the variable agi_dnid? After that I can do a database

I am running AGI script written in C. My script gets triggered but there is no action. I have taken the samplec file from http://home.cogeco.ca/~camstuff/agi.html. it says launched AGI script /var/lib/asterisk/agi-bin/sample.c exec format error . Did I miss anything? I am running asterisk1.0.7 can anyone please help me on these. thanks, Somesh -------------- next part -------------- An HTML

Hi Folks: I used that one example for AGI script on C web, only to fill the working with the Asterisk. I compiled and it worked great. I executed accidentally the ls -l command in directory where was the source and executable, I noted and was surprised that because the executable size was to further 20 times more than source. I executed the gcc -Os source.c -o executable.agi command several

Is there a way of getting the dialled number from an AGI? Is it passed in the initial variables, or can it be pulled out or passed across from the dial plan? Cheers, Ben Merrills Griffin Internet -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040630/71e0bcda/attachment.htm

exten => auth_dial,1,DigitTimeout,5 exten => auth_dial,2,ResponseTimeout,15 exten => auth_dial,3,Read(dialed,IVR/en_enter_destination,0) exten => auth_dial,4,agi(call_start.php|${dialed}) exten => auth_dial,5,dial(SIP/${dialed}@146.82.15.241) I'm trying to get What they dialed put into the PHP script. How do I get the contents of this variable in the php script?

Hi, I am an experienced developer with Windows and familiar with Linux. I am looking for a SIP solution. 1) How does Asterisk compare to VOCAL in terms of support. 2) Is Asterisk free? 3) Where are the docs? Or even better. Where do I start? 4) Will it run on RH9? Thanks in advance. Costas -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

Hello list! I am new in * but i want to learn about its possibilities. I want somebody to tell me if what I want to do is possible with *. I have a teleconference tool which uses SIP and now I am using Asterisk as POTS gateway. When I dial certain number from a telephone I connect with asterisk which asks me for an extension. When I dial certain extension I connect with my SIP application

Hey.. Your saying I can not use flash with ISDN ? What options to I have to transfer a call directly ? ( So I have a free line afterwords) >> What interface are you using? ZapBRI? if so you might be able to do the >> hairpinning if it is supported. Im not using any interface.. But if you know how to do that, let me know and I install that interface. Thx for your answer :) Gr?sse

As you my be aware the X100p cannot collect uk caller id, now I have a modem and a perl script that creates a file /etc/asterisk/callerid.txt on each incoming call in the format number,date,time,name over writing each time a new call comes in I can asterisk read this file and populate the callerid for internal phones and cdr? I think it can be done with AGI but don't know where to start

Ok I have tried to post to this list server but have just gotten the automated reply saying the moderator has to approve it to the list first which was my mistake for sending from the wrong email account. So if the moderator finally approves my questions and you see the same post again "Sorry". My situation is this: I havn't installed Asterisk yet but am curious the general way

Extreme N00b, I am getting the error message "a target does not exist" when running the make install inside the zap directory, probably pretty common, possibly a package I didn't install, just need some insight on it. The same occurs with the libpri and asterisk. -----Original Message----- From: asterisk-users-bounces@lists.digium.com

Forwarded back to the list so others might get the benefit of the answers, and I get fact checked by others. -------- Forwarded Message -------- > From: Lee <leeb00@gmail.com> > Reply-To: Lee <leeb00@gmail.com> > To: Steven Critchfield <critch@basesys.com> > Subject: Re: [Asterisk-Users] udev or not? > Date: Fri, 10 Dec 2004 13:00:29 -0800 > On Fri, 10 Dec 2004

Just change the install prefix directory in asterisk makefile to something like /usr/local/asterisk, and then you will see all the files needed for asterisk to run. > -----Original Message----- > From: Steven Critchfield [mailto:critch at basesys.com] > Sent: Saturday, 1 March 2003 22:03 > To: asterisk-users at lists.digium.com > Subject: Re: [Asterisk-Users] How could I install

Okay, so like this? PHONE1+AD0-SIP/2000 PHONE2+AD0-SIP/3000 PHONE3+AD0-SIP/4000 ALL+AD0AJAB7-PHONE1+AH0AJgAkAHs-PHONE2+AH0AJgAkAHs-PHONE3+AH0- Then you would have Exten +AD0APg- s,1,Dial(+ACQAew-ALL+AH0-,20) Is that right? I have read about the Macros but don't understand their use. Could someone provide an example? Sorry about the newby questions... This will hopefully be my

Hi, this is my 1.st response to this list, i hope this will work. I tend to agree with Steven since just allowing other termination digits probaly wont solve your upcoming the issues anyway. I use a wrapper around the 'get digit' which allows me to specify that the * digit repeats the menu but maxium 3 times and if the * star digit is used twice in sequence (without other digits inbetween

PoE, or 802.3af, uses a device detection routine to determine if the connected device needs power. The process, in greatly simplified terms, is as follows: 1. Detect link state 2. Send a pulse of a known frequency and intensity over the TX/RX pairs 3. Listen for reflection. 3a. No reflection- provide power 3b. Reflection- no power Devices that comply with 802.3af have filters designed

Speaking as former Dialogic/Bayonne user who was frustrated for months with Dialogic's complexity and months of initial testing with GlobalCall only to use their many-years-old and very complex base Dialogic drivers(and eventually scrapping it all for Digium/Asterisk and being up one week later), I can tell you that Digium/Asterisk is definately the way to go. It is 1000 times easier to

Juan, I dont see people here with "attitude" or any need to shut down this list as you suggested. Contrary to your experience, I find people to be quite helpful here. Steven's comment "Start here and then ask a real question" is definitely not arrogant IMHO. He is just trying to knock some etiquette on Luan. Besides, he did give him all the answer Luan was looking for.

Steven Critchfield <critch@basesys.com> wrote: __________ >On Wed, 2004-05-26 at 14:23, tpanton@attglobal.net wrote: >> On which subject, has anyone else >> got time to work with me on a chan_dialogicGC ? It looks do-able but I am ignorant of how asterisk does threading. > >Do you have GPL drivers for the dialogic card? Follow previous comments >about licensing.