Displaying 20 results from an estimated 8000 matches similar to: "Asterisk Java API - Up to date"
2012 May 03
1
AMI disconnects
Hi all.
I've got a perl script that connects to Asterisk's management interface using Asterisk::AMI. So far, its proven to be very useful.
I'm hoping to use this to detect and respond to asterisk restarts and sip reloads.
However, my script gets disconnected quite frequently, causing false alarms in my monitoring.
Here's what the code looks like:
2015 Jan 28
2
queue show <queue-name> vs queue log for calculating average hold time
Hi
We're using 1.8.23.1 on CentOS 5 and are trying to get accurate stats for
queues.
For a particular customer, when I run queue show <queue_name> I get the
following numbers:
<queue_name> has 0 calls (max unlimited) in 'ringall' strategy (17s
holdtime, 94s talktime), W:0, C:175, A:44, SL:48.6% within 45s
So from that data we look at
17s holdtime
And assume that is the
2010 Sep 09
5
info about application not available asterisk 1.6.2.11
Hello list,
how come on my Asterisk 1.6.2.11, I have no help available ?!
asterisk*CLI> core show application Dial
-= Info about application 'Dial' =-
[Synopsis]
Not available
[Description]
Not available
[Syntax]
Not available
[Arguments]
Not available
[See Also]
Not available
Kind regards,
Jonas.
-------------- next part --------------
An HTML attachment was scrubbed...
2012 Sep 26
1
Asterisk 1.8.15.0, Requested transfer capability: 0x00 - SPEECH
Hello,
I'm having issues connecting throu PRI with the following error "Requested
transfer capability: 0x00 - SPEECH"
Below are the logs:
== Using SIP RTP CoS mark 5
-- Executing [97052660 at voipphones:1] Set("SIP/4856-00000003",
"CALLERID(num)=xxxxxxxxx") in new stack
-- Executing [97052660 at voipphones:2] Dial("SIP/4856-00000003",
2010 Oct 21
10
Asterisk 1.80-rc5
Just done a clean install of rc5 on a totally new machine and found the
following:-
/etc/init.d/asterisk start
errors on line 109 - there is no 0 before $VERBOSITY as in the other lines.
More interesting is that after make samples I have no iax2 available.
Dave Cotton
2012 Jun 11
1
Differences between PBX and SBC
Hello,
I would like to know the difference between encrypt the rtp and signaling
between two asterisks, or putting an SBC in front of each Asterisk pbx. I'm
trying to understand whether an SBC could fit an Asterisk deployment
Thanks
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2010 Oct 23
2
Just Take dCAP at Astricon?
Since it is Saturday evening (7PM EST) I am asking this on the list in case
someone who knows sees it and can answer.
Astricon is in my back yard for the first time, and I could hit you with a
rock. I would always like to attend, and spoke at the 2007 Astricon in
Phoenix but don't have the idle cycles.
Question: Can I just go to Astricon and take the dCAP exam only? In and
out? Cost?
I
2010 Jul 24
4
getting some segmentation faults with 1.8
I downloaded the latest 1.8 (27922) but got some segmentation faults.
The first one was when it loaded cdr_odb, and so I changed menuselect
not to compile that one, but the second one was when it tried to load
chan_agent and so I stopped there to see if anyone else was seeing
this. The agents.conf is all commented out except for [general] .
Anyone know what is happening?
Thanks.
P.S. I deleted
2010 Oct 22
5
dials a trunk when off hook
How can I let asterisk immediately dials a trunk when off hook?
2010 Oct 14
6
Audiocodes firmware
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
</head>
<body text="#000000" bgcolor="#ffffff">
<font size="+1">Does anyone have links to the most recent audiocodes
2010 Oct 18
5
Same extension registering over eth0 and eth1
Hello list,
I need to know how to deal with a redundant network with only one asterisk
server, which is receiving registrations from the end points on both of its
ethernet ports. This means extension 201 is registering both from eth0 and
from eth1.
Is there a way/software which can act as a middle man between asterisk and
the ethernet ports, and by default sends registrations to asterisk only
2010 Sep 11
8
No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
Hi Everyone,
I have a provider whose DID used to come into the box just fine but recently
stopped working. Nothing has been changed on our end.
Here is what I get when doing "sip set debug peer PROVIDER":
Sending to 123.123.123.123 : 5060 (no NAT)
^^^^ That is ALL I am getting with sip debug turned on.
With Allow Anonymous SIP set to YES, then the call comes in properly and you
see
2010 Sep 16
4
one way audio for xlite clients behind NAT
I am having a one way audio issue with xlite clients behind NAT. They can
connect to the server and make calls but no audio is heard on the other
end.
my sip conf
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
canreinvite=no[tomfmason]
type=friend
secret=secret
callerid="Thomas Johnson" <XXXX>
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=gsm
2013 Feb 24
3
Asterisk AMI - Create a daemon (background process)
I wanted to create a daemon (background process) in PHP. A daemon will use
socket to connect with Asterisk AMI to send events and listen the actions.
A daemon will also listen the commands from agents via HTTP, for example:
A agent pressed a hang up button on a browser - it will send http command
to a daemon. A daemon received a command and will then send Hang Up Action
to AMI.
How should a
2013 Oct 25
2
Is this big of new modification in Asterisk Events Objects values ?
Hi Team,
Thanks for your great job an Asterisk new features developments. I
installed asterisk-12 Beta and found some changes as well which i notice to
put in-front of your knowledge, don't know that bug of new modification
into objects or old version (asterisk-11) mistake corrected that time,
anyway
*Asterisk-12:*
Array
(
[Event] => ConfbridgeMute
[Privilege] => call,all
[Conference]
2014 Mar 24
1
AMI Proxy
Hi people
Just having a quick check to see if anyone is using any AMI proxies and
which are the most popular. For our purposes it must be able to connect to
multiple asterisk instances.
Thanks for the help.
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: ish at pack-net.co.uk
w: http://www.pack-net.co.uk
Registered
2014 Jul 02
1
Notification when queue member's phone rings
Short question: how to get control or notification (gosub, macro, AGI)
when a queue member's phone starts ringing due to an incoming call from
the queue.
Backround: Our phone operators serve both an asterisk call queue and a
queue for web chat support. I have a gosub on the queue that calls to
our app server to mark the operator unavailable for web chat as soon as
they answer an
2010 Nov 12
3
Official Documentation for Asterisk 1.6 Realtime ODBC Tables
Hi All,
I'm having an issue where Asterisk continuously sends out a GAZILLION "SIP NOTIFY" messages when a user has a voice message in their INBOX. This issue is only present when my SIP users and peers are configured from my ODBC backend (MySQL). A static configuration of users in sip.conf resolves this and everything works fine.
I'd like to confirm the layout of the
2010 Sep 14
9
Speech To Text on linux with asterisk
Hi,
Is it possible to record say 30 seconds of audio and then have LumenVox
convert to text ?
or any available tool open source for speech to text .
Regards
Dhaval
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100914/b56c3d9c/attachment.htm
2010 Oct 24
5
Integrating Asterisk 1.8 with Google Talk and Google Voice
Evening,
Has anyone seen a how-to on getting Asterisk to work with Google Talk
and Google Voice?
Thanks