Displaying 20 results from an estimated 2000 matches similar to: "Dial from AGI = no ring back ??"
2006 Feb 24
0
What's with Indications/SetLanguage/Zaptel/RingBack ?
Good morning everybody,
Can someone explain to me the interconnection between
these four things: indications.conf, SetLanguage(), zaptel.conf
and ring-back ? If there is any !! :- )
I am having this case where some users cannot hear ring back
from a DeadAGI script and it seems to be interconnected to these items.
These users are from the iaxfriends table, they _can_ hear ring-back from
a
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
2008 Jan 12
2
Perl-AGI process
Hi All,
i have created one prepaid PERL AGI script to integrate asterisk users in our current Oracle Billing System. I am using $AGI->exec('Dial', $dialstr); in script after getting the MAX time out for the priticular call.
But when the channels increase on my asterisk more than 50-60 asterisk get crashed and i am suspecting the cause is of AGI Script. because when i check ps on
2009 May 12
1
enum agi interesting problem
Hi,
I am having a strange problem with enum and AGI.
Here is what happens:
I have in my agi something like that:
foreach my $resolver ("e164.arpa", "e164.info", "e164.org") {
my @enums = get_enums($phone, $resolver);
foreach my $enum (@enums) {
$dialstring = $enum .
2006 Jan 27
3
paging agi
Hello Everyone,
I've been playing with an agi script for paging sip phones.
page.agi will take all available sip extensions and assign them to the
global variable PAGE_GROUP. Allowing the phones to be paged from the
dialplan with the new Page cmd. Extensions to be excluded are presented as
arguments to the agi. Each time a page is made this agi refreshes the global
variable. This works with
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
2006 May 23
3
AGI ?
Hi All,
I have been attempting to get an AGI LCRdialout script to work.
Basically what I need to have happen is when someone dials out a number
the script check to see if it is local if so, go out the ZAP channel. If
the ZAP channel is busy, go out the IAX channels, if IAX is all busy, go
out the SIP channels. Here is a sample of what I have in my script.
#!/usr/bin/perl
use strict;
use
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 Jun 24
2
Playing sound before dialing
Hi,
I have configured asterisk now with ENUM lookups which are working
really perfect.
Now I want to play a small soundfile before dial the number to inform
the caller which protocl is used (SIP, IAX2 or ISDN).
How can I do this?
With Playback it doesn't seems to work:
[iax2-sipport-out]
; with leading 3 using IAX-sipport
exten => s,1,NoOp(Dialing ${DIALSTR} with iax2-sipport-out)
exten
2006 Jun 12
2
AGI Stderr
Does anyone know how I can get stderr from AGI to be sent to somewhere other than the console? It seems that this is the only place it can go. Changing logger.conf has no effect.
If you want to see errors from AGI scripts, you have to run the Asterisk console, which isn't viable.
Doug.
2005 Sep 03
1
Current status on _outgoing_ Swedish/Dutch DTMF CLIP for TDM400 FXS interfaces?
Hi all,
I have been looking at the code for both the zaptel driver (wctdm.c/wcfxs.c)
and the asterisk channel driver (chan_zap.c) trying to figure out how much
of this that has been implemented. So far I can see that the current stable
1.0.9.1 zaptel driver don't have the SETPOLARITY ioctl that would be
required to properly signal the Swedish/Dutch CLIP, but the 1.2 beta1 has
this
2006 May 30
1
Asterisk::AGI and DIALEDTIME
Hi List,
In one of my AGIs (using DeadAGI) I grab the answered time using:
my $res = $agi->exec ("DIAL $dialstring");
my $answeredtime = $agi->get_variable ("ANSWEREDTIME");
However this information differs from what's written in the Master.csv
file (which happens to be the correct value!)
Any ideas why?
I'm using asterisk 1.2.7.1 and the
2006 Jun 19
0
Call Not Disconnecting
Hi all,
We are running more than 40 active calls on our
Asterisk Box. But some time we are facing problem,
call is not disconnecting for a long time more than 2
and 2 hrs. in this cuase our customers charged for 1,2
hrs. even they made very small calls.
i have already set rtptimeout = 60, but not
disconnecting
Here is my extentions.
[main-ext]
exten => _x.,1,AGI(main-ext.pl)
exten =>
2010 Oct 10
1
Modifying cid.cid_name in app_parkandannounce.c
Hi List,
I need to modify the callerID name of the call coming back when a parked
call returns to the extension that parked it when it times out.
Looking at app_parkandannounce.c
/* Now place the call to the extention */
snprintf(buf, sizeof(buf), "%d", lot);
memset(&oh, 0, sizeof(oh));
oh.parent_channel = chan;
oh.vars =
2004 Dec 10
0
AGI Perl
Hello,
I'm writing a AGI Perl script and use the following lines to initiate a
call :
$res = $AGI->exec("DIAL $dialstr");
And then :
$answeredtime = $AGI->get_variable("ANSWEREDTIME");
The problem is that I need to know immediately when the call has begun.
I can have the call duration & establishment time at the end of the call
but not in real time when it
2006 Apr 26
3
astcc: need partial pin code
I have not used astcc with pin codes so far, since I set-up the phone
number as card number.
Some of my users want now to dial in to the system and than use their
card, which is their phone number.
For that I would need a way of authentication, like a pin.
I want to use something like:
What is your card number: <user keys in the number>
Enter your pin: <user enter a long pin>
2006 Dec 20
0
asterisk run on vxworks for hardware pbx
Hi
My hardware PBX run asterisk on vxworks,Because the vxworks not support
perl.
Now I want to add a callback function to my pbx.
now it can store Caller and Called party numbers in queue when Called party
is busy
Then I malloc a new ast_channel to call.It is should use
ast_get_channel_by_exten_locked() or ast_channel_alloc() ,
my program as follow,But it isn't work, anyone know how to
2006 Mar 14
3
Attended Transfer - transfer timeout, how to change?
Hi,
We are trying to use attended transfer with Asterisk 1.2.5, but when we
do the transfer and dial the new number, it times out after 3 rings and
then the callee is put back to the original agent.
Where can I adjust the timeout which applies to the number we are
transferring to? I have changed the extension for this number to timeout
at 60 seconds, but that seems to make no difference.
--
2008 Jan 08
4
Bugs??
Good Day All,
I am facing a serious problem since I started to use asterisk. I don?t know if it is a bug or some one already solved this.
Currently I am running 120 VoIP SIP channels on my asterisk server but each day 2, 3 calls got hanged in asterisk, and on asterisk CLI ?show channels? showing us as call UP but in real there is no call.
When asterisk restarted the hanged calls removed from
2004 Dec 21
10
Codec Selection
Hi,
I have 2 g729 licences - what I want to do is use g729 by default but if
I get more than 2 calls at a time, use gsm for the others.
So, I put this on all my sip providers:
disallow=all
allow=g729
allow=gsm
However, this just seems to use gsm for everything. If I comment out the
gsm line, it then uses g729.
I thought it would use the codec's in the order they are allowed - is
this