Displaying 20 results from an estimated 200 matches similar to: "Asterisk Manager"
2007 Jul 08
1
Asterisk Help
Hi
I need help in configuring a auto dialer system using Asterisk. I'm holding
my customers number in MySQL want to fetch 10 numbers one time and dial if
gets connected and answered by customer wants to play a sequence of message
. Please help .
I've tried here is my code to place calls but in this I see no of failure
calls are more than 50%. so please advise.
2011 Feb 13
1
Call Files, Variable passing
Hi,
I am having trouble passing variables via the call files, here is my call
file via the php:
fputs($oSocket, "Action: login\r\n");
fputs($oSocket, "Events: off\r\n");
fputs($oSocket, "Username: $strUser\r\n");
fputs($oSocket, "Secret: $strSecret\r\n\r\n");
fputs($oSocket, "Action: originate\r\n");
fputs($oSocket,
2006 Dec 29
0
PHP to call script
Using the php script below. I am able to enter my number and the number to
call, however I get the following error:
-- AGI Script cid-spoof.agi completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
-- Executing Wait("OutgoingSpoolFailed",
2007 Jan 02
5
Call connected, cannot hear or speak - $20 for fix
I am able to get this script to dial, but I am unable to talk or hear
anything. The script asks for the number to call and the the caller id to
display (if user is not at their normal extension). Once submitted, the
external extension receives a call, once answered the call is then placed to
the dentition number.
The script works as the call is place, but I cannot hear or say anything.
Any one
2007 Jul 08
1
Early Media Handling
Hi
using php script and Asterisk manager I'm dialing numbers and once gets
connected send to an exten in my dial plan that plays an automated message
but some time without answering even it goes to my exten. How can I handle
early media in Asterisk that is I want only when user answer the call it
should goto my specified extension.
my php script:
$oSocket =
2007 Oct 13
0
Set up two PSTN calls and then join them
I wish to set up two PSTN calls and then connect them similar to Jajah (is
this called 3pcc?). The PSTN interconnect is handled by a third party SIP
provider.
I can do this using the manager or call files. An example (using php) would
be:
fputs($oSocket, "Action: login\r\n");
fputs($oSocket, "Events: off\r\n");
fputs($oSocket, "Username: $strUser\r\n");
2007 May 05
2
Manager API Output
Hi,
Is there any way that I can store my manager API output that is:
My question is that is there any why using that I can get the QueueStatus
and store the result in some text file for further processing.
<?php
$strHost = "127.0.0.1";
$strUser = "cron";
$strSecret = "1234";
2012 Dec 12
1
Asterisk 11 originate errors
Hi,
I'm getting errors while originating a call through AMI.
[Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe
[Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe
[Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe
Asterisk version 11.0.1
2009 Dec 18
2
To Asterisk AMI Gurus - Tacking issue with originate
Hello Everyone,
I am making a simple index.php file which will allow a web user to enter his
$phoneNumb, $dialNumb, and callerID ($spoofNumb) and get the call bridged.
Following is the index.php and the contents of extensions_custom.conf. When
I submit the form nothing happens. I don't even see Manager Connected msg.
Your input will be much appreciated. I am thinking I have some syntax
2009 Dec 23
1
AMI originate and PHP
Hi Guys,
I am trying to make a web form where a person is allowed to put in
$phoneNumber, $dialNumber, and $spoofNumber to make a call with spoof caller
ID. There are a few problems that I am facing with Asterisk AMI Originate
command. The reason why I want to use the darn AMI Originate is because I am
sending calls to mobile phones and I want to have some accountability and to
know if a call was
2009 Oct 02
0
Sending a DTMF remotely with PlayDTMF problem.
Hello,
I need to be able to send a DTMF to an existing channel remotely. So I made
a php script to do such with the Manager command PlayDTMF. I need it for
example to start a transfer.
isb177*CLI> features show
Builtin Feature Default Current
--------------- ------- -------
Pickup *8 *8
Blind Transfer # #8
Attended Transfer
2009 Oct 03
0
Problem sending a DTMF remotely. Please need help...
Hello,
I need to be able to send a DTMF to an existing channel remotely. So I made
a php script to do such with the Manager command PlayDTMF. I need it for
example to start a transfer.
isb177*CLI> features show
Builtin Feature Default Current
--------------- ------- -------
Pickup *8 *8
Blind Transfer # #8
Attended Transfer
2009 Oct 06
0
Problem sending a DTMF remotely. Please need help!!!
Hello, how are you?
I need to be able to send a DTMF to an existing channel remotely. So I made
a php script to do such with the Manager command PlayDTMF. I need it for
example to start a transfer.
isb177*CLI> features show
Builtin Feature Default Current
--------------- ------- -------
Pickup *8 *8
Blind Transfer # #8
Attended
2009 Oct 05
1
Problem sending a DTMF remotely. Please need help!!
Hello,
I need to be able to send a DTMF to an existing channel remotely. So I made
a php script to do such with the Manager command PlayDTMF. I need it for
example to start a transfer.
isb177*CLI> features show
Builtin Feature Default Current
--------------- ------- -------
Pickup *8 *8
Blind Transfer # #8
Attended Transfer
2013 Feb 23
0
click2call with AMI?
Hi,
I have a PHP code with AMI to using in click2call system.
here is my code:
$user = "usernamr";
$secret = "secret";
$channel = 'SIP/' . $sip;
$context = "from-internal";
$waitTime = "20";
$timeout = 20000;
$priority = "1";
$maxRetry = "2";
$pos = strpos($number,
2007 Jun 05
1
IAX2 Trunk No Sound
Hi
I've two boxes connected over IAX2 trunk before IAX I was using SIP trunk
and they were working fine b'coz of bandwidth issue I changed from SIP to
IAX now I'm facing a strange problem after some time on the cli of my
asterisk box I see lots of messages of IAX2 trunk and b'coz of that my
agents are not able to hear any thing and I've restart my * box. Please
guide me what I
2007 Jun 04
2
G729 License
HI
I bought 20 license from Digium and install in my server and b'coz of some
problem I've to change my server is it possible that I can use those lice
and register again in my new server ?
Is it possible that I'll be able to use those lice in my old box also ?
thanks
arun
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2014 Mar 04
0
Anybody have issues joining machines to a S4 AD domain using VBS
All
got an issue I have a VBS that should join machines to the domain but it's
not working get access denied instantly
anybody got any working scripts they would like to share with me.
btw my script that doesn't work is below
' Join a Computer to a Domain
Const JOIN_DOMAIN = 1
Const ACCT_CREATE = 2
Const ACCT_DELETE = 4
Const WIN9X_UPGRADE = 16
Const DOMAIN_JOIN_IF_JOINED =
2020 Mar 11
0
[PATCH RFC v2 12/24] hpsa: use reserved commands
On Wed, Mar 11, 2020 at 12:25:38AM +0800, John Garry wrote:
> From: Hannes Reinecke <hare at suse.com>
>
> Enable the use of reserved commands, and drop the hand-crafted
> command allocation.
>
> Signed-off-by: Hannes Reinecke <hare at suse.com>
> ---
> drivers/scsi/hpsa.c | 147 ++++++++++++++------------------------------
> drivers/scsi/hpsa.h | 1 -
2007 Jun 04
0
IAX2 Trunk Problem
Hi
I've two boxes connected over IAX2 trunk but suddenly my cli is getting
flood with these messages:
iax2_trunk_queue: Maximum data space exceeded
and b'coz of that my agents are not able to hear any thing.
when this happened that time there were 9 calls.
my * version is 1.2.18 and 1.2.14
thanks
arun
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