Displaying 20 results from an estimated 1000 matches similar to: "Getting Exec Format Error when running AGI call"
2006 Jan 16
2
AGI variables
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
2006 May 15
1
GET DATA and STREAM FILE commands, don´t work
Hi,
I have been written an small script for test the use these commands. I had done massive test with commands, but I didn?t get success
it. Any of the cases, I don?t listen nothing on channel that call 2100 extension. I dial 2100 extension through an cisco phone 7912 with SIP, also I dialed through ATA SIP (Linksys PAP-2).
I?m using Asterisk 1.2.7.1 and ztdummy driver, linux kernel 2.6.11.4. I
2006 Dec 12
1
AGI problema
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type">
<title></title>
</head>
<body bgcolor="#ffffff" text="#000000">
<font face="Verdana">Hi all. I've written a AGI in C language.
2008 Jan 27
1
[AGI 1.4] C sample?
Hello
I'm pretty much a newbie when it comes to C, but I have to use
this language to write a couple of AGI proggies because I need them to
be statically compiled.
Strangely enough, Google didn't return much when looking for the
"Hello, world!" of AGI in C.
The following doesn't work: The file never gets written:
===========
//check_cid.c
#include <stdio.h>
#include
2011 Feb 22
1
[1.4.39.1/AGI] ast_carefulwrite: write() returned error: Broken pipe
Hello
Incoming calls from the FXO trigger an AGI script which simply NOOP
data sent by Asterisk through stdin.
The first two NOOP work fine, but after this, Asterisk isn't happy:
============= extensions.conf
...
[from_fxo]
exten => s,1,Wait(2)
exten => s,n,Set(CID=${CALLERID(num)})
exten => s,n,AGI(/var/tmp/test.lua)
exten => s,n,Wait(5)
exten => s,n,Hangup
=============
2012 Jan 12
1
how to set callerid in php AGI file.
Hi,
I am using phpagi for agi scripting. and want to update callerid number but
didn't get any success. please help me how to update PHPAGI is new for me.
Below is the code which I write.
#!/usr/bin/php -q
<?php
set_time_limit(30);
//require(.phpagi.php.);
include("phpagi.php");
$agi = new AGI();
//answer the call
$agi-> answer();
2011 Feb 24
1
missing argument on AGI
Hi All,
I'm using the asterisk 1.4.39.2 with phpagi 2.20 I have setup a dial plan:
[callback-outbound]
exten => _00.,1,Macro(callout|${EXTEN})
[macro-callout]
exten => s,1,AGI(getchannel.php|${ARG1})
exten => s,2,Dial(Local/${OUTBOUND}@from-internal/nj||tr)
exten => s,3,Hangup()
but for some reason i am not receiving the argument:
Executing [s at macro-callout:2]
2005 Jun 11
1
AreskiCC Calling Problem
Hello There,
I *think* i've setuped the AreskiCC2 Calling Card system right , but
i've yet to make any calls out of it , i added a rate card , trunk
and defined some rates , generated some users , added 10 dollars in
them , okay , now i call any number , it asks me to enter my pin , i
do , it tells me i have ten $ , right after that it says sorry you
dont have enough funds for this call
2005 Oct 06
1
Fwd: ASTCC - INUSE Flag
Hi all. Just to update list and increase the "souls-save" database.
The patch solved the problem. Now I have an asterisk-1.2.0-beta1 with
asterisk-perl-0.08 and mysql-server-3.23.58-16.FC2.1 machine working
fine with ASTCC and "inuse" flag.
The link of the patch is: http://bugs.digium.com/view.php?id=5400
Best regards to all you in the list.
Ricardo Poppi.
2011 Feb 24
2
[1.4] Still can't get it to call back
Hello
No matter what I try, Asterisk still fails dialing back through a
callfile built through an AGI script.
The whole thing works fine when the original call that triggers
Asterisk is from an internal extension (Xlite), but it fails when it's
from my cellphone ringing through the FXO/Zaptel port and I want to
wait a few seconds and call back through the FXO/Zaptel.
Could it that even
2005 Sep 28
3
ASTCC - INUSE Flag
I download and installed ASTCC over the weekend and I am having an issue where the INUSE flag will not get set back to 0 if the user drops a call while the balance is being played. All other times it seems to reset the flag correctly.
I have tried both AGI and DeadAGI with the same results.
Those of you using it for a while, how did you get around this?
Just for fun this is all I am doing in
2005 Sep 22
0
ASTCC error when using silent=5
Hi list. I?m using ASTCC with callerid authentication and got things
working fine, except for one single issue:
Using this command -- DeadAGI(astcc.agi|${CALLERIDNUM}|${EXTEN:1}|5) --
and passing the 5 parameter for silent, it exits unexpectedly. I tested
the 4, 3, 2 and 1 and they are working ok.
But... when I use the five it returns zero and exit without executing
the dial command! Take a
2007 Jul 10
0
Odd AGI Issue - STREAM FILE, GET DATA not playing file
Apologies if this has been brought up before, but extensive googling
and digging through my list archive didn't turn anything up.
Basically, I'm working on an AGI web app and need to read some digit
input. I'm having multiple issues with asterisk interpreting agi
commands at the moment, but I figured I'd start with this one.
when I call GET DATA or STREAM FILE I don't
2008 Oct 17
0
GET DATA Returning only a single digit
--
jand. more than just a group
Asterisk AGI Command GET DATA is usually of this form
GET DATA timeout max_digits
When I execute this command, I get only a single digit, regardless of
what the value of max_digits is,
Also the script quits Immediately after the press of the digit
regardless of what the value of timeout is,
This is really un-desirable as I will like to GET multiple DTMF digits
2008 Feb 06
0
Problem forwarding a call with an AGI script
Hi,
I'm trying to achieve the following:
Incoming call for user A (97), user A make a blind transfer to user B's
phone (96).
User B's phone rings and since there is no one to take the call, it
returns the call to User A with an AGI script.
The dialplan looks like this:
[local]
....
exten => 96,1,Dial(SIP/user4,10,tr)
exten => 96,2,AGI(transfer.php)
exten =>
2011 Feb 24
2
[1.4.39.2] Simple AGI doesn't reply
Hello
The following, dead simple Bash script ran as AGI doesn't reply to
Asterisk:
============= extensions.conf
[from_fxo]
exten => s,1,Wait(2)
exten => s,n,Set(CID=${CALLERID(num)})
exten => s,n,AGI(/var/tmp/basic.agi)
exten => s,n,Hangup()
============= /var/tmp/basic.agi
#!/bin/bash
#Ripped from
#http://lists.digium.com/pipermail/asterisk-users/2003-July/008554.html
while
2008 Feb 04
8
AGI: Not getting answers from get_data in a call-file call
I have the following situation: I drop a call-file into the Asterisk
spool directory and I get called back. That all works.
And I have this script:
#!/usr/bin/perl -w
use Asterisk::AGI;
my $AGI = new Asterisk::AGI;
my %input = $AGI->ReadParse();
$AGI->answer();
my $i;
$i = $AGI->channel_status();
$AGI->say_digits($i);
$i =
2007 Aug 09
1
generating a GUID
I have a need to have a GUID (for example,
bcd47ccc-d7c9-ddb6-dc11-6746a770d77d [36 characters long including the
"-"]) generated in the dialplan. Is there any asterisk function that
would do this ? I would prefer not to have to shell out every time a
call comes in.
Julian
2011 Dec 15
1
Wrong call information on B leg
Greetings.
I have next feature in features.conf :
send =>
*9,peer/both,AGI,/etc/asterisk/agi/map_mail.pl
What it does is parsing
CALLERID and DNID from AGI input, performing some actions in MySQL with
these values, and then running application for peer (for example,
PlayBack)
Sounds simple, and it really is. When my user is receiving a
call (we are the B leg) and presses *9, everything
2009 Jun 04
2
broken pipe in perl agi
Hi gang,
Since I'm getting no joy from device_Status or SIPPEER in
1.4.26-rc1, I thought I would do an AGI to read my hints and check for line
in use that way. The AGI works fine from a prompt, but returns the dreaded
"utils.c:966 ast_carefulwrite: write() returned error: Broken pipe" when I
try to run it from the dialplan. Here is my dialplan snippet;