Displaying 20 results from an estimated 600 matches similar to: "[1.4.39.1/AGI] ast_carefulwrite: write() returned error: Broken pipe"
2011 Mar 11
1
Asterisk 1.8 AGI error ast_carefulwrite: write() returned error
Hey Guys,
We upgrade asterisk from 1.2.x to 1.8.2.3 and my one of agi script doesn't working We have allpage.agi script for paging system on all polycom 501 but after upgrade it broke. Any idea what is this error ?
extension.conf
exten => 7770,1,agi(allpage.agi)
exten => 7770,2,meetme(7770,dq)
exten => 7770,3,playback(beep)
exten => 7770,4,hangup
following is agi debug....
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;
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
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
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
2009 Feb 26
3
Getting SIP field P-Asserted-Identity from EAGI
Hi, using EAGI variables like
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
I get a lot of data about a call, but I need to obtain P-Asserted-Identity
value from a SIP call. Are tehe any eagi variable to get that? Or have you
any solution??
Thanks!!!
-------------- next part
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
$agi-> answer();
2006 Dec 12
1
AGI problema
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type">
<title></title>
</head>
<body bgcolor="#ffffff" text="#000000">
<font face="Verdana">Hi all. I've written a AGI in C language.
2005 Sep 28
3
ASTCC - INUSE Flag
I download and installed ASTCC over the weekend and I am having an issue where the INUSE flag will not get set back to 0 if the user drops a call while the balance is being played. All other times it seems to reset the flag correctly.
I have tried both AGI and DeadAGI with the same results.
Those of you using it for a while, how did you get around this?
Just for fun this is all I am doing in
2008 Jan 27
1
[AGI 1.4] C sample?
Hello
I'm pretty much a newbie when it comes to C, but I have to use
this language to write a couple of AGI proggies because I need them to
be statically compiled.
Strangely enough, Google didn't return much when looking for the
"Hello, world!" of AGI in C.
The following doesn't work: The file never gets written:
===========
//check_cid.c
#include <stdio.h>
#include
2008 Mar 24
2
Getting Exec Format Error when running AGI call
Dear friends,
I am having problem with running a sample php and I can't figure out why. I
can run the sample.php using CLI but when I run it inside the dialplan it
does not work. Can someone please suggest the config problem that I may
have made?
dommy:/var/lib/asterisk/agi-bin# php sample.php
#!/usr/bin/php5 -q
VERBOSE "Here we go!" 2
VERBOSE "Call from - Calling
2011 Feb 24
1
missing argument on AGI
Hi All,
I'm using the asterisk 1.4.39.2 with phpagi 2.20 I have setup a dial plan:
[callback-outbound]
exten => _00.,1,Macro(callout|${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]
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
2005 Jun 03
1
ARESKICC DOESN'T make a CALL!!!
Hi Folks,
After going to the paifull steps of installing AreskiCC and finally being able
to access the webinterface, connecting to *, importing rates and setting up
accounts I am not being able to make a CALL: No matter what number i try to
dial I get the same response: The number you have dialed is currently
unavailabel. Please enter thenumber you want to dial starting with 1 for
local and
2011 Feb 27
1
[Dahdi 2.4.0] Flash() hangs up
Hello
I need Asterisk (1.4.39.2) to simulate a flash hook (ie. hitting the
"R" key on European handsets) so I can put a call on hold, dial a
second number, and set up a conference call.
By default, linux/include/dahdi/kernel.h sets the flashtime to 750ms,
which appears to be too long for European telcos, as they seem to
expect a line cut of about 100ms.
After editing the
2009 Jul 26
0
after 1.4.26 upgrade: "ast_carefulwrite: write() returned error: Broken pipe"
Hi,
After upgrading a debian/lenny server to 1.4.26 I get this error:
== Manager 'munin' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'munin' logged on from 127.0.0.1
[Jul 26 17:45:12] ERROR[12354]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe
repeated each time munin logs in.
Should I be concerned?
2003 Nov 10
3
AGI and PHP
i've just spent the pass 2 days trying to get AGI to work with PHP;
i made a lot of silly mistakes along the way which could have been
avoided if only there were some kinda howto or samples. at the risk
of looking stupid, i decided to shared my experience in hopes that
it might help some newbie get going with PHP.
1. first order of business is to be aware of your php environment; i m
NOT
2009 Oct 22
2
carefulwrite: write() returned error: Broken pipe
Dear,
I am getting this in CLI on release candidate version of Asterisk. Any
ideas, or points where to look?
-- Launched AGI Script /var/lib/asterisk/agi-bin/rad-auth.agi
[Oct 22 18:21:45] ERROR[9853]: utils.c:1126 ast_carefulwrite: write()
returned error: Broken pipe
-- <SIP/916-fc001968>AGI Script rad-auth.agi completed, returning 0
Best regards,
Josip
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