Store the call count in a shared SQL db.
Sent from my Verizon Wireless 4G LTE DROID
Brian LaVallee b.laval...@globaltank.jp wrote:
I would like to enforce call-limit across multiple servers. Is there
any way to pass a call-limit variable between servers 01 02, as shown
below? Use a global
That's just disgusting If you want to run your phones on WindBlows
use lync Should be plenty point and click easy for you
On 12/04/2013 09:19 AM, CDR wrote:
Digium is 100% lost in the map. If they would come up with a Paid
version of Asterisk, one that would use the .NET framework
I second that!
Sent from my Verizon Wireless 4G LTE DROID
Eric Wieling ewiel...@nyigc.com wrote:
Asterisk is Open Source, any company can port Asterisk to Windows.Nobody
has. Personally, I don't want Digium taking valuable and limited development
resources to create a Windows port.
Enable the full log in logger.conf. then in asterisk CLI run sip set debug on.
Wait for it to happen again, then check the log file /var/log/asterisk/full to
see if there's a sip statement being sent to your server to enable music on hold
Sent from my Verizon Wireless 4G LTE DROID
Dominik
Hey All,
Growing call center. Currently at about 200 call center staff, running
about 1000 calls per hour. Gearing up to double that. Not too sure that
a single server will support that growth. So, I'm trying to come up with
ways to scale the system and still maintain a simplistic design. So
).
It is based on this idea:
http://www.opensips.org/Documentation/Tutorials-LoadBalancing-1-9
- Laszlo
2013/8/27 Gregory Malsack gmals...@coastalacq.com
mailto:gmals...@coastalacq.com
Hey All,
Growing call center. Currently at about 200 call center staff,
running about 1000
interface that connected to the cel database that they could use to find
it for themselves. With the cel_pgsql mod crapping out all the time, I
feel like that was a waste of my time
Thanks! Look forward to your input...
Gregory Malsack
Ok now for the meat
Asterisk Server
Dell 1950 Dual Quad
I have a Dell 1950 dual quad core 2.5 croon server running 215 phones and 110
concurrent calls. 16GB ram. We record every call to an nfs share, and CDR to a
separate mySQL server.
Greg
gmals...@coastalacq.com
bilal ghayyad bilmar...@yahoo.com wrote:
Hello;
If I have load up to 220
*2.5 ghz. Gotta love auto correct. Additionally, total volume its about 1200
calls per hour.
Greg
Gregory Malsack gmals...@coastalacq.com wrote:
I have a Dell 1950 dual quad core 2.5 croon server running 215 phones and 110
concurrent calls. 16GB ram. We record every call to an nfs share
I've used a lot of Dlink DES-1228p and 1210-28p. Primarily with polycom phones.
Seem to have pretty good luck with them for the last 7 years or so.
Eric Wieling ewiel...@nyigc.com wrote:
Adtran
-Original Message-
From: asterisk-users-boun...@lists.digium.com
No. Although Nicolas may have gone on holiday. I just purchased 2 licenses for
fop2 a month or so ago.
Carlos Alvarez car...@televolve.com wrote:
We have licensed both products and sent a support request on 6/11, with
zero reply or any activity on it at all so far. No replies to subsequent
I just took the dCap exam Friday. Good things to study is to take the dcaa exam
online. If you can breeze thought that, you should be in good shape for the
written. As for the practical exam. You'll want to make sure you know
endpoints, dahdi, and trunking.
However, I also took the advanced
I think it runs off the OS time...
Joseph syscon...@gmail.com wrote:
Which file in Asterisk have a setting for time zone?
When asterisk record incoming call in Master.csv the time is 6hr. ahead.
I'm on: Canada/Mountain zone
--
Joseph
--
/etc/asterisk/cdr.conf
Joseph syscon...@gmail.com wrote:
On 05/11/13 20:44, Gregory Malsack wrote:
I think it runs off the OS time...
That what my impression was but all my records written to:
/var/log/asterisk/cdr-csv/Master.csv
are 6hr. ahead; so they are based on UTC and no correction
Matt,
At some point you need to consider how much is too much...
I run a call center with more then 125 commissioned phone sales reps and more
than 60 customer service reps. We run dual servers, fiber from one provider and
6 bonded T1's from another provider. We purchase our so trunks from a
Hello All,
History ~
I recently took a position with a call center. At the time they had
about 50 agents in a call queue. The queue was setup to ringall. The
agents use Eyebeam softphones. Everything is local lan, no routers,
everything connected via Cisco 3600 10/100 switches.
Now we are
Asterisk version 1.8.20.1
Already checked the switches, no noteworthy port issues. no vlans used
or layer 3 switching.
On 03/28/2013 03:18 PM, Carlos Alvarez wrote:
On Thu, Mar 28, 2013 at 12:55 PM, Gregory Malsack
gmals...@coastalacq.com mailto:gmals...@coastalacq.com wrote
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] inbound call issue...
insecure=very should fix it.
On Wed, Nov 3, 2010 at 4:08 AM, Gregory Malsack gmals...@gmellc.com wrote:
Can anyone tell me why my inbound calls keep getting rejected with 401?
Here’s the debug
Can anyone tell me why my inbound calls keep getting rejected with 401?
Here's the debug information:
--- SIP read from UDP:147.135.32.221:5060 ---
INVITE sip:6087294...@216.26.109.22:5060 SIP/2.0
Call-ID: 31007e...@147.135.32.221
CSeq: 1 INVITE
From: Wi
Does anyone have any first-hand experience with the Zoiper Business version
softphone? If so what has been your experience with it?
Thanks,
Greg
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Hello All,
Is there anyone out there that is able to integrate a custom visual basic 6
application to Fonality’s Trixbox HUD Pro?
Thanks,
Greg
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asterisk-users mailing
Hello All,
I have a need to connect an analog device to an asterisk server. The analog
device has 4 analog lines going into it (it’s a fax solution). The fax solution
answers the analog call, then listens for dtmf. The dtmf code that is played
tells the fax device what email address to send
I would like to thank everyone for their input. This project is completed and
the solution is working wonderfully! There was some mention of some people
having difficulties with asterisk/mssql connectivity via the dialplan when
under heavy load. Running the connection through agi was the
Hello Everyone,
I have an installation where the client has a Microsoft SQL database that holds
all of their case information. They would like the asterisk system to require
users to enter a valid case number when making an outgoing call. I’m seeing
some documentation regarding people using
...@lists.digium.com] On Behalf Of Gregory Malsack
Sent: Thursday, December 18, 2008 19:37
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Authorize Microsoft SQL
Hello Everyone,
I have an installation where the client has a Microsoft SQL database that holds
all of their case
This much I already know. This information is easily found through a simple
google search. What I'm looking for is if anyone knows what a dialplan would
look like that would perform an ODBC query to an ODBC database. I've seen
minuet documentation on ODBCget, which is what I'm thinking will do
Of Gregory
Malsack
Sent: Thursday, December 18, 2008 20:13
To: f...@teamforrest.com; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Authorize Microsoft SQL
This much I already know. This information is easily found through a
simple google search. What I'm
Hey Everyone,
Here’s an email I received from a client who has a trixbox system that has
contracted with me for some custom dialplan programming.
While I was away at a conference on Tuesday, our server crashed same as before
(it was “responsive”
Hello Everyone,
I’ve sold an asterisk system to a client that has a custom written CRM package
written in VB6 with an MS-SQL backend. They want to “unplug” the application
from their old phone system and “plug” it into the asterisk system. The program
has pop-up screens based on incoming
Hi All,
Can anyone recommend a good VOIP provider in the Milwaukee/Chicago area? We
need flat rate billing per line/trunk, trunking, did’s, and iax or G.729
compatibility.
Thanks,
Greg
No virus found in this outgoing message.
Checked by AVG.
Version: 7.5.524 / Virus Database:
, but he wants a sequence that he can push, so
that when he rants and raves at a customer, there won't be evidence to say that
he did that... :)
Just a hunch on that. :)
I don't know.
Eugen
On 7/22/08, Gregory Malsack HYPERLINK mailto:[EMAIL PROTECTED][EMAIL
PROTECTED
stopmixmonitor. However I am unable to locate examples of syntax on that
command. Here is what I have:
stoprecording = *8,self/callee,StopMixMonitor,
This command syntax does not work and the recording continues on. Can anyone
provide direction on this?
Thanks,
Gregory Malsack
-Original
channels simultaneously, the recording is only assigned to 1 channel and you
have to run the command against the originating channel of the call.
Greg
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Malsack
Sent: Wednesday, July 23, 2008 2:13 PM
Hello,
My boss is asking me to setup the asterisk server to record all calls.
(Simple). However, he wants to be able to enter a key sequence during the call
to stop the recording. Any ideas on how I would do that?
Thanks,
Greg
No virus found in this outgoing message.
Checked by AVG.
Does anyone know of a good VOIP dialtone provider in the northern Chicago area.
My client has tried Broadvoice and Mix and is having problems with latency in
the middle of the traceroute between him and the provider.
Thanks,
Greg
No virus found in this outgoing message.
Checked by AVG
That is correct; we would not recommend using just *any* CF card, as the
write speed of the card needs to be pretty high to be able support
multiple voicemail messages being written simultaneously. With that
said, though, it is possible to use a higher capacity CF card, but my
previous response
Yea, sounds like they've planned for this issue. Kevin, is there an sdk that
can be used to create our own binaries should we want to add modular support
for something? Like say mysql cdr's?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian J.
,
Gregory Malsack
President
Classic Services
Select Digium Reseller
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming
Sent: Saturday, December 29, 2007 7:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk
and delete, or destroy
all copies of this message immediately.
_
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Malsack
Sent: Monday, December 17, 2007 11:36 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Queue calls drop to voicemail intermittantly
That is the same thing I thought as well. However the queue is set to an 18
second timeout and the voicemail is set to a 20 second timeout. I'll increase
the voicemail timeout so there is a little more play there just to see if that
helps.
-Original Message-
From: [EMAIL PROTECTED]
] On Behalf Of Tilghman Lesher
Sent: Monday, December 17, 2007 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queue calls drop to voicemail intermittantly
On Monday 17 December 2007 12:35, Gregory Malsack wrote:
Can anyone tell me what might cause callers
The grammar makes it hard to understand the question, but if I’m understanding
this right, this will probably to the trick.
In the queue config file add:
member = Agent/(agent’s id number)
to the end of the queue directives. Otherwise if you are trying to say that you
want the agent
Can anyone tell me what might cause callers on hold in a queue to drop
into agents voicemail boxes?
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To UNSUBSCRIBE or update options visit:
Actually I believe the process you are describing is the agentcallback feature.
Once you are logged in if the agent is configured to have voicemail and does
the light should come on.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen
Sent:
will erratically
drop calls to the queue.
Any help will be extremely appreciated, and I will provide any conf
files you may require. I have included excerpts of the config files I
think you may need.
Sincerely,
Gregory Malsack
Incoming line in extensions.conf
exten = 8582294,1,answer()
exten
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