have your answer.
On Wed, Jul 23, 2014 at 4:03 PM, Eduardo Leones
edua...@ypytecnologia.com.br wrote:
Thanks for the feedback.
In this case SSD disks you think it solves?
Eduardo
2014-07-23 18:01 GMT-03:00 Ron Wheeler rwhee...@artifact-software.com:
I would also do some math
Another question, what audio format I use in MixMonitor to maintain a
connection with reasonable quality and reduce the use of I / O disk? Today
I use wav.
tks
2014-07-24 9:05 GMT-03:00 Eduardo Leones edua...@ypytecnologia.com.br:
Thank you all for the answers. I will do tests to find
people
I have a running Asterisk 1.8.28 in great Dell server with two xeon
processors and 16gb of ram and HD SAS 15k (Raid 1). This server is
recording all calls (placed to record the audio in a ram disk), the entire
CDR goes straight to MySQL by cdr_mysql.so. Each call runs some validation
and
the resource usage indicators and drive busy/throughput statistics.
On Wed, Jul 23, 2014 at 2:48 PM, Eduardo Leones
edua...@ypytecnologia.com.br wrote:
people
I have a running Asterisk 1.8.28 in great Dell server with two xeon
processors and 16gb of ram and HD SAS 15k (Raid 1
to increase the concurrent calls, or use a storage appliance.
To confirm this, install the tool nmon and use the v and d options to
bring up the resource usage indicators and drive busy/throughput statistics.
On Wed, Jul 23, 2014 at 2:48 PM, Eduardo Leones
edua...@ypytecnologia.com.br wrote
Hello,
I'm running Asterisk on a CentOS 64-bit server. . Asterisk if I compile
using the ./configure --libdir=/usr/lib64 instead of ./configure have a
relative gain performace.? Has anyone done any comparison?
Is there any way in the compilation or even in settings that I can improve
the
Hello,
My question is this, I have a service queue that members follow the service
interval (wrapuptime = 30).
However, sometimes these members need to call the customer back, thus
making an active call. Occurs when this member disconnects the call shortly
following section in the queue already
Guys, I have a problem. I have a queue on asterisk 1.8 that members are
added dynamically via the AMI QueueAdd. When you run the CLI a
reload app_queue.so all members who were in the queue disappear. This is
a bug or some parameter that I do not know?
Would have another way to do the reload queue
Josh, thanks for the feedback. That problem can also occur with dynamic
members, would not be just for those who work with realtime?
tks
2014-06-06 10:14 GMT-03:00 Josh Metzger joshdmetz...@gmail.com:
On Fri, Jun 6, 2014 at 9:03 AM, Eduardo Leones
edua...@ypytecnologia.com.br wrote
Hello!
I wonder what the default context that the Queue application uses to call
extensions. If there is a possibility to change this into a context created
by me possible? Would you like to get this load value to variables before
calling the extension.
tks,
Eduardo
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Hello,
How do I track the status of an extension for socket? I'm trying to use the
ExtensionState, but it is returning empty.
thank you,
Eduardo
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Hello, working in a call center where we set up a structure in asterisk.
When my voip reaches 150 calls are with bad quality. We do not transcode
codec. What I realized using the top command server (CentOS) processing is
too high for the asterisk. But the general processor server is down. Would
I have a question about global variables. Is it possible to somehow keep
global variables unset via Dial Plan even Restarting asterisk?
tks
Eduardo
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New
minutes just to ask for the password of
the agent.
I thought I might be missing hardware, but both the CPU and Memory are with
low consumption.
Does anyone have any idea what happens to the AgentCallBackLogin?
tks,
Eduardo Leones
Good afternoon,
I'm not able to create a dial plan to pull links extension. I am
using asterisk 1.4.18with the following dial plan:
exten = _ * 7XXX, 1, Pickup ($ {EXTEN: 2} @ PICKUPMARK)
exten = _ * 7XXX, n, Hangup ()
But is not working, the following error appears in the CLI:
-
Assunto: Re: [asterisk-users] pickup for extension in asterisk 1.4?
On 11-09-03 02:19 PM, Eduardo Leones wrote:
Good afternoon,
I'm not able to create a dial plan to pull links extension. I am using
asterisk 1.4.18with the following dial plan:
exten = _ * 7XXX, 1, Pickup ($ {EXTEN: 2
List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Enviadas: Sábado, 3 de Setembro de 2011 16:56
Assunto: Re: [asterisk-users] pickup for extension in asterisk 1.4?
On Sat, Sep 03, 2011 at 11:19:12AM -0700, Eduardo Leones wrote:
I'm not able?to create?a?dial plan?to pull?links
10%
De: Matt Riddell li...@venturevoip.com
Para: asterisk-users@lists.digium.com
Enviadas: Quarta-feira, 4 de Maio de 2011 0:32:28
Assunto: Re: [asterisk-users] Fading voice problem
On 3/05/11 10:16 PM, Eduardo Leones wrote:
Guys,
I'm having problems
Anyone know a good IAX2 softphone for Windows that has g729 and it is free?
att
Eduardo--
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New to Asterisk? Join us for a live introductory webinar every
Good morning ...
I'm using Asterisk 1.4.40 AgentCallbackLogin in a Call
Center. What is happening isthat when the Call Center has more than
15 simultaneous calls the login application isextremely slow to fall
into the low priority, ie, the agent can log in, but takes about 1minute
to drop in
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