Displaying 20 results from an estimated 20000 matches similar to: "Proper AGI use with MySQL"
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 Oct 07
1
ASTCC -- semantic note of 'callstart' in cdrs?
Looking at the code, it would appear that the 'callstart' column of the cdrs table should really be
called 'callend':
$dialstr = "IAX2/$res->{path}/$phone|30|HL(" . ($maxtime * 60 * 1000) .
":60000:30000)";
$res = $AGI->exec("DIAL $dialstr");
$answeredtime =
2006 Nov 02
1
AGI Problems
Hi,
I've got a setup whereby calls come into the asterisk server (1.2.7.1)
over a IAX2 trunk and into a dialplan that launches a php AGI script:
[live-full]
exten => _X.,1,Set(TIMEOUT(absolute)=0)
exten => _X.,2,NoOp(${EXTEN})
exten => _X.,3,DEADAGI(live-full.php)
exten => _X.,4,Wait,2
exten => _X.,5,Hangup
The script is using phpagi-2 from http://phpagi.sourceforge.net/ and
2006 Feb 02
1
DeadAGI variables confusion
Hi * users,
We're using calls to external scripts through AGI at various points
throughout our IVR system. We use these scripts to log certain events
and to make certain choices that I wasn't sure would be possible in
the dial plan. The problem comes with with the final call to our
script. We use this line:
exten => h,1,deadagi(log.php|{$service}|Hung up|${UNIQUEID})
I know there are
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 =
2011 Apr 09
1
Is it the normal behaviore for AGI and DeadAGI to terminate AFTER the "h" extension?
Hi Everyone,
Trying to run a php script after DeadAGI for A2Billing does it's magic. This
is the dialplan:
[a2billing]
exten => _X.,1,System(php pre-call.php ${CALLERID(num)} ${EXTEN}
${UNIQUEID})
exten => _X.,n,AGI(a2billing.php,1)
exten => _X.,n,Hangup()
*exten => h,1,Wait(5)*
*exten => h,n,System(php post-call.php ${CALLERID(num)} ${UNIQUEID})*
As you can see above, I even
2009 Mar 09
3
problem with an agi in PHP
Hello,
I need to execute an agi in php.
I have that:
== Using SIP RTP CoS mark 5
-- Executing [0170725000 at mnupprx1:1] Answer("SIP/33179977999-b6c18478",
"") in new stack
-- Executing [0170725000 at mnupprx1:2] GotoIf("SIP/33179977999-b6c18478",
"0?6:3)") in new stack
-- Goto (mnupprx1,0170725000,3)
-- Executing
2007 Sep 15
2
AGI/PHP: missing arguments
hi folks,
I've built a simple PHP-script utilizing the AGI-interface. in
extensions.conf I trigger the script and pass a single value as first
argument:
exten => h,1,DeadAGI(process.php|${Enter})
On the Asterisk-console, I can actually see that the script is called
correctly (something like "DeadAGI(process.php|1234)"). However, when I read
stdin in the PHP script, I receive
2006 Feb 27
8
AGI Scripts Terminate too Soon
Ok, here's a weird one.
I have an AGI script where one user calls another. The call is answered. Everything is peachy. If the call is terminated by the CALLEE hanging up the call, then Asterisk returns control back to where the Dial() command left off, and I can check the return code of Dial(), ${DIALSTATUS} etc. That's all great.
HOWEVER, if the CALLER hangs up the call, it seems as if
2005 Jul 29
1
FastAGI problems
Hello!
I use FastAGI very frequently [meaning 30 channels IVR is made in it]
and sometimes I find, that there is a message like:
Jul 29 09:38:55 VERBOSE[896] logger.c: == Auto fallthrough, channel
'Local/1@route-out-eeae,2' status is 'CHANUNAVAIL'
Jul 29 09:38:55 VERBOSE[893] logger.c: > Channel
Local/1@route-out-eeae,1 was never answered.
Jul 29 09:38:55 VERBOSE[896]
2005 May 11
1
HELP: ASTCC (AGI) meets call forward ERROR
Hi, ALL:
When I use astcc to do the prepaid function, but if I want to enable
"call forward".
The result of CDR seems not correct.
UA 1011 make a call to UA 9999, and UA 9999 forwards this call to a PSTN number.
I think we shall charge the credit from UA 9999 not UA 1011 because UA
1011 don't know where UA 9999 forwards to.
But in CDR, I can only find the from(1011) and
2006 Mar 20
4
simple perl-agi - where's the error?
Hello!
I'm trying to setup a perl-deadagi, but my perl skills lack. can
someone tell me why the following code doesn't work:
#!/usr/bin/perl
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
$dialstring = $AGI->get_variable("DIALSTRING");
$res = $AGI->exec("DIAL $dialstring");
the asterisk output says:
AGI Rx << GET VARIABLE DIALSTRING
AGI Tx >> 200
2007 Dec 07
2
PHP AGI script
I've got a very nice PHP AGI script but I want to be able to do some
database cleanup when the user hangs up the phone. I wish everyone would
hang up when they were suposed to, but some people don't. So what does
Asterisk send to an AGI file when the line has been disconnected? If I
remember reading somewhere correctly, I don't need to use DeadAGI. Instead
I'm able to use
2005 Jan 13
1
problems with astcc
hello *'s,
Astcc not workin what is correct format for defining
1-database
2-brands
3-trunks
4-routes
i define all these things but not workin may be i define in wrong
format.I have FXO card installed.can anyone implement it and also my
sip phone generates very loud noise wat is that i tried several
settings but not hear any voice just noise.
sip.conf
[general]
context=from-sip
port=5060
2007 Aug 13
1
AGI answering the channel even though I never asked it to
I am working on a call-back solution where the initiating call should
never be answered.
I was doing this simply through the dial plan, sending a progress
tone, and then dumping the channel, and firing off a DeadAGI which
created a call file to make the callback.
Now I've tried extending this so that an AGI is fired first to check
for things - like no inbound ANI - and play a
2005 Mar 15
2
How to determine the voicemail file name for an AGI script
I've read several of the Wiki sections on Voicemail and "Asterisk
variables" but could not find an obvious answer to this question.
I would like to run a script that post-processes the voicemail after the
Voicemail application returns (with AGI or DeadAGI), but I cannot figure
out how to easily determine the name of the file written by the
Voicemail application. Does anyone
2010 Feb 14
3
Line DC
My dialer works perfectly , but whenever I dial a number manually from xlite
and press a Key like 6055 for DTMF , line gets disconnected. Line gets DC as
soon as I press any key from xlite
What could be the issues ?
I tried the SAME VOIP from another center and Its Ok there.
I tried the Same dialer Xlite over Static IP, problem is there.
I tried the same number from other Dialer , it works
2010 Aug 11
2
channel variables in AGI
Hello,
How to take the values of channel variables like 'agi_uniqueid' and
'agi_callerid' in agi script.
For example
#!/bin/bash -x
T="$agi_uniqueid"
I want to save value of 'agi_uniqueid' channel variable into a variable
called 'T' in my script
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2006 Feb 20
1
Dial from AGI = no ring back ??
Hi everybody,
I sent an e-mail this morning regarding SIP / IAX2
with no ring-back, I now succeeded to pin-point the
problem, here it is, if I dial a provider directly from
extensions.conf I get ring-back, if I dial from an AGI
script I don't get the ring-back but it calls anyway.
I use 1.0.9.
Any hint would be appreciated ! Thanks,
Frederic
;Calling this one does not give me ring back
2007 Jun 15
2
combining AGI with dialplans
On 2007-05-15 Tony Mountifield wrote (wrt using AGI scripts to dial out):
> Can't comment on this one, as I never use AGI to dial.
> My AGIs just set the context, extension and priority,
> and exit to the dialplan to do any dialling.
(http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/185537)
I would like to do this, but I am having trouble figuring out how. I have