Displaying 20 results from an estimated 9000 matches similar to: "Limit on simultaneous manager commands"
2005 Jan 04
0
Manager API - ExtensionState help please.
I'm not having any luck getting the ExtensionState action of the Manager
API to work.
The response is always success but the Status is always -1 which to me
means an error.
Here is a typical telnet session.
-----------------------------------
stockholm:~ # telnet localhost 5038
Trying ::1...
telnet: connect to address ::1: Connection refused
Trying 127.0.0.1...
Connected to localhost.
Escape
2009 May 20
2
Manager ExtensionState function
Hi,
I am trying to get the extension status (weather it has dialed
outgoing call via SIP or IAX2), using the following piece of code
however it always returns -1 on all the extensions (valid/invalid).
Am i missing something ? Any help.
Thanks
-----------------------------------
#!/usr/bin/perl
use Asterisk::Manager;
use lib './lib', '../lib';
$|++;
my $astman = new
2005 Jan 13
4
Manager API !!!!!!!!!
Hello all
Has anyone had any success with the Manager API ?
I am trying to check an extension status without too much luck I have
the following
<?php
$fp = fsockopen("127.0.0.1", 5038, $errno, $errstr, 30);
if (!$fp) {
echo "$errstr ($errno)<br />\n";
} else {
$out = "Action: Login\r\n";
$out .=
2005 Feb 02
0
ExtensionState problems using Manager.conf API
This is my first attempt to write software of any sort. What I am
trying to is to use a .php page to query asterisk Manager and get the
ExtensionState for each particular extension. Then when it has the
answer it outputs an XML file for use as the directory on a Cisco 7960
phone. What I am thinking is that when the user hits the directory
button to veiw the directory that is at this URL it will
2008 Jun 28
0
AMI extenstion state
Hi,
I would like to get the status of asterisk extension with my php program.
*My program as follows,*
<html>
<!--<meta http-equiv="refresh" 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
2009 Jan 16
0
No subject
different stand alone linux server which act as my routers. Here is a
picture showing the output from the CISCO switch going to the two
linux servers:
http://www.grmtech.com/blog/wp-content/uploads/2009/02/cisco2950-24ports-farleft-two-output-300x89.jpg
My questions are:
1. The black wire coming into the Mc Manstel box is that a fibre optic cable ?
2. What is the Mc Manstel box doing ?
3. What
2007 Feb 10
1
SIP retry time too low
I have a problem with asterisk-1.2.13, where it retries SIP INVITEs
too quickly. It happens when qualify is on, and the server it tries to
reach is only 1ms away according to qualify.
The time between the first SIP INVITE and the 7th (last) is then only
64ms, and that can be too short for the peer to react.
I reported this bug in much more detail in bugs.digium.com, but the
bug is gone now
2009 Jan 06
1
"username mismatch, have <x>, digest has <y>"
I have two Asterisks connected using SIP. One is acting as a SIP
"server", the other as a SIP "client". This almost works; but calls
from 50607795 are rejected with this error:
check_auth: username mismatch, have <50607796>, digest has <50607795>
On the "client" I have these accounts configured in sip.conf:
register => 50607795:test at
2009 Aug 02
1
T.38 and reinvite
I have a setup with a number of customer Asterisks with T.38 enabled.
This works quite well for each customer sending faxes between branch
offices.
They all have a SIP trunk to a central Asterisk, which connects them to
the PSTN through various providers on dedicated lines. I cannot enable
reinvite on those SIP trunks, because that would allow calls from the
customer's phones to get
2009 Oct 14
2
Queues with unavailable members
We have the possibly rather unique setup where we have cell phones
posing as SIP devices. The SIP registration for those unfortunately
doesn't go away just because the phone is off, since the registration is
done by our cell-phone<=>SIP gateway, and that gateway has no way of
knowing whether the phone is on or off.
This is usually ok, but it gets problematic if the cell phone is a
2003 Dec 28
0
Is there something wrong with "show manager commands"?
Is it just my box, or is there something flaky in the implementation of
"show manager commands"?
Note: I'm using putty. About half way through this, I toggled my KVM over to
the desktop and logged in to try and recreate it. The output was the same as
the last two entries in this dump.
bebop*CLI> show manager commands
bebop*CLPing Ping
bebop*CLLogoff Logoff Manager
2010 Nov 18
2
exceeds the maximum size of ast_fdset error on Asterisk-1.8.0
Hi Friends,
i have installed and configure asterisk-1.8.0.
When i have tried asterisk start get below errors and not able to start
asterisk.
*FD 32767 exceeds the maximum size of ast_fdset!*
Thanks in advance.
--
Best Regards,
Rajnikant Vanza
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2005 Sep 15
0
AW: ***SPAM*** actionID on manager events
hi,
afaik, the action-id provided with the OriginateAction should only show up in the OriginateSuccess or OriginateFailure event. Intermediate events that are generated when the channels are create will NOT carry the action-id of the originate.
The async flag tells asterisk to process originates in parallel, i.e. if you have two users originating calls and NO async flag set, the second originate
2008 Feb 01
7
Enterprise or Fedora?
i wanna build a production Asterisk box ,will RedHat Linux Enterprise Server be more stable than Fedora core Linux or it makes no significant difference
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2009 Jul 01
2
Testing the manager.conf: sending and receiving commands
Hi All;
How can I test manager.conf?
Can I telnet to the asterisk machine at the port 5038 and send and receive commands to test if the manager is working fine? How?
Regards
Bilal
2005 Sep 13
2
actionID on manager events
Hello, all!
I'm looking at the wiki page and info on the mailing list and I'm getting
conflicting info...
I am using the manager API from the telnet CLI and I am testing creating calls
with it. I login with events: on and I can originate calls just fine.
However, when I set ActionID on an Originate, I cannot see anywhere where that
actionid carries into the Event output.
But I found
2009 Jul 22
3
ExecIf and empty variables (early evaluation)
Imagine that you have this code:
exten => _X!,n,Set(foo=${QUEUE_WAITING_COUNT(${QueueName})}))
If ${QueueName} happens to be unset, this will cause a warning:
[Jul 22 14:26:17] ERROR[8114]: app_queue.c:5187
queue_function_queuewaitingcount: QUEUE_WAITING_COUNT requires an
argument: queuename
The obvious solution:
exten => _X!,n,ExecIf($["${QueueName}" !=
2008 Jan 07
3
How to check if a SIP phone is forwarded without ringing it ?
Hi,
I feel I've read a thread about this previously but I couldn't find it.
Is there way for an Asterisk server to check if a sip phone is forwarded
without bothering phone's user ?
I was thinking of some Alert-Info option that would let the phone reply with
a 302 Moved Temporarily or 182 Queued message and not let the phone ring or
display anything on its screen.
So that, you could
2012 Jun 05
3
CDRs on multiple servers.
Hello guys,
I need to be able to throw cdrs on more than one servers at a time. Please let me know how this can be done.
Thanks
2006 Oct 10
2
E164 caller ID
Is there a proper and accepted way to go about setting an E164 compliant
caller ID (ANI) ?
Currently, we're using just the Set(CALLERID(num)=XX) where XX is some E164
compliant number like 3539146632431 or some such.
Is there another way we should be doing that or is that proper?
N.