Displaying 8 results from an estimated 8 matches for "amp111".
2006 May 26
2
Asterisk.NET authentication problem
...am called Asterisk.NET.Test and it
uses the following default constants for login:
const int ASTERISK_PORT = 5038;
const string ASTERISK_HOST = "10.34.9.206";
const string ASTERISK_LOGINNAME = "admin";
const string ASTERISK_LOGINPWD = "amp111";
However, when the application tries to login using these constants I get
an "Authentication Failed" message.
In /var/log/asterisk/full log:
May 26 08:06:33 DEBUG[28367] manager.c: Manager received command 'Login'
May 26 08:06:33 VERBOSE[28367] logger.c: == Par...
2006 Jan 27
3
paging agi
...print "VERBOSE \"Net::Telnet NOT INSTALLED - this is required\"
0\n";
exit 0;
}
# You need to configure this: Your manager API username and password. This
# is the information from /etc/asterisk/manager.conf. You need something
like
# this in it:
# [admin]
# secret = amp111
# deny=0.0.0.0/0.0.0.0
# permit=127.0.0.0/255.0.0.0
# read = system,call,log,verbose,command,agent,user
# write = system,call,log,verbose,command,agent,user
# IF that's what you have in your conf file, this is what you should have
here:
my $mgruser = "admin";
my $mgrpass = "amp1...
2005 Feb 02
1
Reproducible crash with CVS stable (from about 5 days ago...) - but only from iax clients
...ster@digium.com>
=========================================================================
Connected to Asterisk CVS-v1-0-01/28/05-15:21:35 currently running on
centrala (pid = 28749)
-- Remote UNIX connection
Verbosity was 3 and is now 11
centrala*CLI> MANAGER LOGIN MD5 127.0.0.1, admin, amp111
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'admin' logged on from 127.0.0.1
-- Executing Queue("SIP/201-ec33", "prodaja") in new stack
-- Started music on hold, class 'default', on SIP/201-ec33
-- Stopped music on hold on SI...
2005 Jul 13
1
Polycoms and paging
I'm looking at deploying some Polycom 501's here, but one thing that still
needs confirmation before I can move forward is global paging.
I figure that I can couple polycom auto-answer
(http://www.voip-info.org/tiki-index.php?page=Polycom+auto-answer+config) with
this script:
http://lists.digium.com/pipermail/asterisk-users/2004-March/040186.html
However, that script was posted over a
2008 Jun 28
0
AMI extenstion state
...sh" content="1" />-->
<?php
$fp = fsockopen("xxx.xxx.xxx.xxx", 5038, $errno, $errstr, 30);
if (!$fp)
{
echo "$errstr ($errno)<br />\n";
}
else
{
$out = "Action: Login\r\n";
$out .= "UserName: admin\r\n";
$out .= "Secret: amp111\r\n\r\n";
fwrite($fp, $out);
$in = "Action: ExtensionState\r\n";
$in .= "Exten: 777\r\n\r\n";
$in .= "Context: ext-did-custom\r\n\r\n";
$in .= "ActionID: 1\r\n";
//$in.= "Status: State\r\n";
$in .= "Action: Logoff\r\n\r\n\r\n";
fwr...
2006 Dec 29
0
PHP to call script
...";
#specify the username you want to login with (these users are defined in
/etc/asterisk/manager.conf)
#this user is the default AAH AMP user; you shouldn't need to change, if
you're using AAH.
$strUser = "admin";
#specify the password for the above user
$strSecret = "amp111";
####-----This block is not used in this script----###
#specify the channel (extension) you want to receive the call requests with
#e.g. SIP/XXX, IAX2/XXXX, ZAP/XXXX, etc
#$strChannel = "IAX2/XXXXXX";
#specify the context to make the outgoing call from. By default, AAH uses
from...
2005 Sep 09
2
AMP 1.10.009 released!
Hello all,
Asterisk Management Portal 1.10.009 has now been released. This
exciting new version has several notable additions (listed below).
The AMP homepage is http://amp.coalescentsystems.ca. Here you'll find
links to the download, install guide, and documentation wiki.
As usual, please use amportal-users mailing list for discussions about
AMP:
2007 Jan 02
5
Call connected, cannot hear or speak - $20 for fix
...";
#specify the username you want to login with (these users are defined in
/etc/asterisk/manager.conf)
#this user is the default AAH AMP user; you shouldn't need to change, if
you're using AAH.
$strUser = "admin";
#specify the password for the above user
$strSecret = "amp111";
#specify the channel (extension) you want to receive the call requests with
#e.g. SIP/XXX, IAX2/XXXX, ZAP/XXXX, etc
$strChannel = "Local/15555555555@outrt-001-telasip";
#specify the context to make the outgoing call from. By default, AAH uses
from-internal
#Using from-internal w...