Re: [asterisk-users] implementing call center using asterisk

[Sylvain Boily]

Le 2016-06-22 12:47, Goke Aruna a écrit :

hello all,
I am looking for an implementation of a 10 man call center. low cost 
license or GPL will be preferred.

I will be glad for your help.
Regards




Hello Goke,

XiVO has call center features (it's GPL) : 
http://documentation.xivo.io/en/stable/contact_center/contact_center.html


Sylvain
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] implementing call center using asterisk

[Goke Aruna]

Thanks Carlos,
Have you used any of them?
Regards

On Wed, Jun 22, 2016 at 6:32 PM, Carlos Rojas  wrote:
> Hi
>
> You can use, gnudialer, vicidial, goautodial.
>
>
>
>
> On Wed, Jun 22, 2016 at 12:47 PM, Goke Aruna  wrote:
>>
>> hello all,
>> I am looking for an implementation of a 10 man call center.  low cost
>> license or GPL will be preferred.
>> I will be glad for your help.
>> Regards
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] implementing call center using asterisk

[Carlos Rojas]

Hi

You can use, gnudialer, vicidial, goautodial.




On Wed, Jun 22, 2016 at 12:47 PM, Goke Aruna  wrote:

> hello all,
> I am looking for an implementation of a 10 man call center.  low cost
> license or GPL will be preferred.
> I will be glad for your help.
> Regards
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] implementing call center using asterisk

[Goke Aruna]

hello all,
I am looking for an implementation of a 10 man call center.  low cost
license or GPL will be preferred.
I will be glad for your help.
Regards
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Collaboration Call Center Integrated with Asterisk web and email

[Lenz Emilitri]

A number of call-centers I see use the pause codes in Asterisk to mark
different types of activities, like answering to email or IM. It's not
much, but easy to implement.
l.


2012/3/27 bilal ghayyad bilmar...@yahoo.com

 Hi All;

 Is there a collaboration contact center (hope to be open source)
 Integrated with Asterisk (hope with vicidial), so the agent will be able to
 receive chat or emails sessions and deal with the customer. If the agent in
 a call with the customer, then he will not get chat session. Is there like
 this software?

 Regards
 Bilal

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




-- 
Loway - home of QueueMetrics - http://queuemetrics.com
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Collaboration Call Center Integrated with Asterisk web and email

[bilal ghayyad]

Hi All;

Is there a collaboration contact center (hope to be open source) Integrated 
with Asterisk (hope with vicidial), so the agent will be able to receive chat 
or emails sessions and deal with the customer. If the agent in a call with the 
customer, then he will not get chat session. Is there like this software?

Regards
Bilal

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asternic Call Center and Asterisk 1.4 Queues

[JR Richardson]

Hi All,

I'm testing the Asterinic Call Center Queue Log Analizer.  Working ok
except for realtime monitoring.  The page updates queue summary and
calls waiting, but not Agent status.  When an agent is (busy) in
[asterisk queue show], the 'state' of the agent in agent status on
the web page does not change, always shows 'not in use'.  The page
does update with 'Last In Call' info after hangup of a call.

Any ideas?

Thanks.

JR
-- 
JR Richardson
Engineering for the Masses

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Remote Call Center Agents and Asterisk?

[d4rk f1br]

Anyone using Asterisk in a Call Center environment?  And more importantly is
anyone supporting home based remote call center agents with an Asterisk
backend?

My experience with Asterisk is limited, however I have set it up and
installed it previously and had it working for home usage and for simply
playing around.  My background however is with Cisco CallManager, Cisco
IPCCX for call centers as well as a mixed bag of other big name systems.

I am simply researching and investigating different possibilites and
solutions for a project at this point.  Pursuing as many avenues as possible
and trying to setup various test beds and labs if you will to accomplish the
goal of one day rolling out home based remote call center agents.

Look forward to hearing from others about this.  Looking to hear of any
success stories, as well not so successful stories.  Trials and
tribulations, good and bad experiences and where that left you.  I know
others have at the least done exactly what I am doing and have researched
and entertained various ideas regarding this model of home based agents so
hopefully this message can be a catalyst for further disscussion around this
trend.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Remote Call Center Agents and Asterisk?

[Steve Totaro]

On Feb 1, 2008 10:33 AM, d4rk f1br [EMAIL PROTECTED] wrote:
 Anyone using Asterisk in a Call Center environment?  And more importantly is
 anyone supporting home based remote call center agents with an Asterisk
 backend?

 My experience with Asterisk is limited, however I have set it up and
 installed it previously and had it working for home usage and for simply
 playing around.  My background however is with Cisco CallManager, Cisco
 IPCCX for call centers as well as a mixed bag of other big name systems.

 I am simply researching and investigating different possibilites and
 solutions for a project at this point.  Pursuing as many avenues as possible
 and trying to setup various test beds and labs if you will to accomplish the
 goal of one day rolling out home based remote call center agents.

 Look forward to hearing from others about this.  Looking to hear of any
 success stories, as well not so successful stories.  Trials and
 tribulations, good and bad experiences and where that left you.  I know
 others have at the least done exactly what I am doing and have researched
 and entertained various ideas regarding this model of home based agents so
 hopefully this message can be a catalyst for further disscussion around this
 trend.


I have had several very successful implementations.  Some small
~50-100 agents, and some larger, around 500 agents.

The trick is keeping the agents honest since there is no supervisor
standing behind them.  You want to establish a minimum standard for
the home agent.

Recording of calls for sound quality and agent evaluation will be
critical for QA as well as ChanSpy and whisper coaching which I
believe is available in 1.4 (?)

Use AJAX and Jabber ActiveX controls to control your CRM (web based of course).

You could even ship them a Kit which contains a router running
DD-WRT or OpenWRT setup with an OpenVPN tunnel back to your data
center, an ATA, and a headset.

There are several benefits to sending the router.  Obviously, VPN.
Then you also have something to SSH into and do testing like ping,
traceroute, test throughput.  You could even create some kind of app
(if it doesn't already exist) to regularly run these diagnostics and
upload them to you.

That is just a few ideas but I think the main thing you will run into
is agents not logging off or somehow trying to beat the system.  That
is what I see time and time again.

Thanks,
Steve Totaro

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Remote Call Center Agents and Asterisk?

[Steve Totaro]

On Feb 1, 2008 10:33 AM, d4rk f1br [EMAIL PROTECTED] wrote:
 Anyone using Asterisk in a Call Center environment?  And more importantly is
 anyone supporting home based remote call center agents with an Asterisk
 backend?

 My experience with Asterisk is limited, however I have set it up and
 installed it previously and had it working for home usage and for simply
 playing around.  My background however is with Cisco CallManager, Cisco
 IPCCX for call centers as well as a mixed bag of other big name systems.

 I am simply researching and investigating different possibilites and
 solutions for a project at this point.  Pursuing as many avenues as possible
 and trying to setup various test beds and labs if you will to accomplish the
 goal of one day rolling out home based remote call center agents.

 Look forward to hearing from others about this.  Looking to hear of any
 success stories, as well not so successful stories.  Trials and
 tribulations, good and bad experiences and where that left you.  I know
 others have at the least done exactly what I am doing and have researched
 and entertained various ideas regarding this model of home based agents so
 hopefully this message can be a catalyst for further disscussion around this
 trend.


I have had several very successful implementations.  Some small
~50-100 agents, and some larger, around 500 agents.

The trick is keeping the agents honest since there is no supervisor
standing behind them.  You want to establish a minimum standard for
the home agent.

Recording of calls for sound quality and agent evaluation will be
critical for QA as well as ChanSpy and whisper coaching which I
believe is available in 1.4 (?)

Use AJAX and Jabber ActiveX controls to control your CRM (web based of course).

You could even ship them a Kit which contains a router running
DD-WRT or OpenWRT setup with an OpenVPN tunnel back to your data
center, an ATA, and a headset.

There are several benefits to sending the router.  Obviously, VPN.
Then you also have something to SSH into and do testing like ping,
traceroute, test throughput.  You could even create some kind of app
(if it doesn't already exist) to regularly run these diagnostics and
upload them to you.

That is just a few ideas but I think the main thing you will run into
is agents not logging off or somehow trying to beat the system.  That
is what I see time and time again.

Thanks,
Steve Totaro

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Remote Call Center Agents and Asterisk?

[EdPimentl]

Hello Steve,

You are right on track and this is also what we have done with pretty good
results.
Of course now with Flex/Air there are a number of ways to enhance the
service for the
Customer/Agent

Ed

Mail:   edpimentl[at]gmail.com
Voip:   edpimentl [SKype | GoogleTalk ]

http://agileoss.com (Web2.0 and SOA Development )
http://mobiquity.ws (Private Label Social Network)
http://youbiquity.ws (Power of One for all Social Networks)
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Re: Call center manager for Asterisk (Release 0.3)

[nik600]

just to let you know that i've started a mailing list on sourceforge

[EMAIL PROTECTED]

You can subscribe here
https://lists.sourceforge.net/lists/listinfo/ccmanager-users

Other news regarding ccmanager will be posted on this mailing list, i
invite interested people to subscribe.

Thanks

On 3/14/07, nik600 [EMAIL PROTECTED] wrote:

Hi

i just want to let you know that is available a new release of ccmanager.

I've added the possibility to import queue_log information in a mysql
database and to generate reports using this information.

The software is in a beta state and provides this functionality:

- users management
- call generation (making a GET or POST request on a certain URL)
- queue management (LOGIN / LOGOUT / QUEUE STATUS)
- pickup a call from a queue even if the user isn't logged in the queue
- outbound call in customizable context
- queue stats import from queue_log
- queue reports creation (using an open xml format)

Please note, i think that the xml definition of a report is very
important, if many people share each other their reports there is the
possibility to build a reports-repository, so the final user can use
many reports and, if the user know what he is doing, he can customize
the reports.

I am looking for people to improve this project, any help would be appreciated.

- developers (php / mysql / postgres / ajax )
- tester
- graphics (div  css)

Here there are some screenshots

https://sourceforge.net/dbimage.php?id=115442
https://sourceforge.net/dbimage.php?id=115440
https://sourceforge.net/dbimage.php?id=114381

And here there is the sourceforge project.

https://sourceforge.net/projects/ccmanager

Thanks, nik


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Mini call center only 15 seats fxs to sip suggestion

[Erick Perez]

Hi,
I looking for an affordable (maybe used) FXS to SIP media gateway (or
another method) to be deployed in a mini call center.
The final user already has analog phones and a cabling setup in place.
The cheap gateway will send and receive SIP traffic to an asterisk box
that is already in place and connected to PSTN.
The asterisk is there because it will provide voice recording and
voicemail to email and a simple IVR.
The final user does not want to spend the money associated with items
like and audiocodes gateway or a sngoma remora or digium FXS card.
that's why we are looking for a media gateway. Since he already have
some analog panasonic phones, he does not want to purchase Ip phones.

if you have some other ideas, let me know.

Ebay turned nothing in my searches.

Thanks,


--

Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Multitenant Call Center Setup

[Lenz]

Hello Waldo,
if you use AddQueueMember plus a fake queue_log registration, you can tell  
who the agent was, not just from what terminal she was connecting from. It  
is then possible to report who was available at a certain time, or see  
agents logging on and off, going to pause, measuring the average call  
length per agent, pause time, etc. If you run a call center you will want  
these pieces of information, otherwise you have no means of understanding  
what is going on. Simply connecting to terminals is not good, because  
you'll usually have more agents than terminals (to compensate for shifts,  
sick leave, vacations, etc)

Bye
l.


On Thu, 20 Oct 2005 17:41:31 +0200, Waldo Rubinstein [EMAIL PROTECTED]  
wrote:



I have played with AddQueueMember and it works great. However, there
is one problem that I have and I hope someone can point me in the
right direction.

My client's agents rotate seats. This means that if I want to track
calls by agent, I can't with AddQueueMember. When I look at the CDR,
it tells me the calls made/received by the station (regardless of
technology - SIP/AIX/etc). But, at any given point, I don't know
which agent made the call.

In reality, even with AgentCallBackLogin I can't tell which agent
made or received the call. Is there a way that I can identify in the
CDR which agent actually received or placed a call regardless of
which extension he/she may be sitting on?

Thanks,
Waldo




--
Loway Research - Home of QueueMetrics
http://queuemetrics.loway.it
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Multitenant Call Center Setup

[Waldo Rubinstein]

Lenz,

Thanks for the response. I agree with you. However, I have a couple  
of questions:


1) How to do a fake queue_log registration
2) One of the needs I have is to be able to generate the calls  
received or made by an agent in real time. I figured I could do this  
by querying the CDR, but I was wondering how to flag the calls made/ 
received by the agent in such a way that I could identify them in the  
CDR. I guess I could always write a script that would insert queue  
events into a database and then I could cross query the CDR with the  
queue_log tables to figure out which was the last agent logged in on  
a particular station and match that against the dstchannel CDR  
column, but that could be hairy.


In order to address #2, I thought of doing the following (I haven't  
tried this yet):


1) Create an extension the agent would dial to log in. This extension  
could look something like:


1,AGI(get_valid_agent)
2,GotoIf($[ ${AGENT} =  ]?9:6)
3,DBPut(agents/${CHANNEL}=${AGENT})
4,AddQueueMember(queue_name)
5,Playback(agent-loginok)
6,Hangup()

where the AGI script will prompt the user to enter an agent id which  
it could verify against a database for this tenant, which would then  
allow me to have multiple agents 1001 (for example) for different  
tenants. CHANNEL would be a cleaned version of the regular CHANNEL  
variable with only the technology and peer name, for example, only  
SIP/1234 instead of SIP/1234-2bc7 and AGENT would be the agent id  
validated by the AGI script, for example, 1001.


2) Say my inbound calls are routed to an incoming context, I would  
add this to the incoming context

[incoming]
...
exten = h,1,DBGet(AGENT=agents/${BRIDGEPEER})
exten = h,2,SetAccount(${AGENT})

where BRIDGEPEER would be a cleaned version of the regular BRIDGEPEER  
variable with only the technology and peer name, for example, only  
SIP/1234 instead of SIP/1234-6a67.


This would allow me to set the Account Code in the CDR to the agent  
logged in using that station for calls distributed by the Queue  
application.


3) In the contexts that allow my agents to make outbound calls, I  
would add something like:


DBGet(AGENT=agents/${CHANNEL})
SetAccount(${AGENT})

where CHANNEL would be a cleaned version of the regular CHANNEL  
variable with only the technology and peer name, for example, only  
SIP/1234 instead of SIP/1234-92b3


This would allow me to set the Account Code in the CDR to the agent  
logged in using that station whenever outbound calls are made on that  
extension. I could even deny the station from placing outbound calls  
if AGENT is blank, meaning that no agent has logged in yet.


4) Finally, I would create an extension for an agent to log out, that  
could look like this:


RemoveQueueMember(queue_name)
DBDel(agents/${CHANNEL})
Playback(agent-loggedoff)
Hangup()

where CHANNEL would be a cleaned version of the regular CHANNEL  
variable with only the technology and peer name, for example, only  
SIP/1234 instead of SIP/1234-1f89


This would remove the entry in astDB at the same time it removed the  
station from the queue.


I haven't tried any of it but I think it could work.

Thanks,
Waldo

On Oct 21, 2005, at 3:39 AM, Lenz wrote:



Hello Waldo,
if you use AddQueueMember plus a fake queue_log registration, you  
can tell who the agent was, not just from what terminal she was  
connecting from. It is then possible to report who was available at  
a certain time, or see agents logging on and off, going to pause,  
measuring the average call length per agent, pause time, etc. If  
you run a call center you will want these pieces of information,  
otherwise you have no means of understanding what is going on.  
Simply connecting to terminals is not good, because you'll usually  
have more agents than terminals (to compensate for shifts, sick  
leave, vacations, etc)

Bye
l.


On Thu, 20 Oct 2005 17:41:31 +0200, Waldo Rubinstein  
[EMAIL PROTECTED] wrote:




I have played with AddQueueMember and it works great. However, there
is one problem that I have and I hope someone can point me in the
right direction.

My client's agents rotate seats. This means that if I want to track
calls by agent, I can't with AddQueueMember. When I look at the CDR,
it tells me the calls made/received by the station (regardless of
technology - SIP/AIX/etc). But, at any given point, I don't know
which agent made the call.

In reality, even with AgentCallBackLogin I can't tell which agent
made or received the call. Is there a way that I can identify in the
CDR which agent actually received or placed a call regardless of
which extension he/she may be sitting on?

Thanks,
Waldo





--
Loway Research - Home of QueueMetrics
http://queuemetrics.loway.it
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update 

Re: [Asterisk-Users] Multitenant Call Center Setup

[Lenz]

Hi Waldo,
about how to create fake queue_log entries, the answer is quite simple:  
see http://www.oinko.net/astrecipes/index.php?n=100
I instead doubt that you can use the CDR for real-time logging, as CDR  
data usually gets written when the call ends. Of course you can hack with  
it, but this is not likely the best way to go. :-)

Bye
l.





On Fri, 21 Oct 2005 10:23:55 +0200, Waldo Rubinstein [EMAIL PROTECTED]  
wrote:



Lenz,

Thanks for the response. I agree with you. However, I have a couple of  
questions:


1) How to do a fake queue_log registration
2) One of the needs I have is to be able to generate the calls received  
or made by an agent in real time. I figured I could do this by querying  
the CDR, but I was wondering how to flag the calls made/received by  
the agent in such a way that I could identify them in the CDR. I guess I  
could always write a script that would insert queue events into a  
database and then I could cross query the CDR with the queue_log tables  
to figure out which was the last agent logged in on a particular station  
and match that against the dstchannel CDR column, but that could be  
hairy.


In order to address #2, I thought of doing the following (I haven't  
tried this yet):


1) Create an extension the agent would dial to log in. This extension  
could look something like:


1,AGI(get_valid_agent)
2,GotoIf($[ ${AGENT} =  ]?9:6)
3,DBPut(agents/${CHANNEL}=${AGENT})
4,AddQueueMember(queue_name)
5,Playback(agent-loginok)
6,Hangup()

where the AGI script will prompt the user to enter an agent id which it  
could verify against a database for this tenant, which would then allow  
me to have multiple agents 1001 (for example) for different tenants.  
CHANNEL would be a cleaned version of the regular CHANNEL variable with  
only the technology and peer name, for example, only SIP/1234 instead of  
SIP/1234-2bc7 and AGENT would be the agent id validated by the AGI  
script, for example, 1001.


2) Say my inbound calls are routed to an incoming context, I would add  
this to the incoming context

[incoming]
...
exten = h,1,DBGet(AGENT=agents/${BRIDGEPEER})
exten = h,2,SetAccount(${AGENT})

where BRIDGEPEER would be a cleaned version of the regular BRIDGEPEER  
variable with only the technology and peer name, for example, only  
SIP/1234 instead of SIP/1234-6a67.


This would allow me to set the Account Code in the CDR to the agent  
logged in using that station for calls distributed by the Queue  
application.


3) In the contexts that allow my agents to make outbound calls, I would  
add something like:


DBGet(AGENT=agents/${CHANNEL})
SetAccount(${AGENT})

where CHANNEL would be a cleaned version of the regular CHANNEL variable  
with only the technology and peer name, for example, only SIP/1234  
instead of SIP/1234-92b3


This would allow me to set the Account Code in the CDR to the agent  
logged in using that station whenever outbound calls are made on that  
extension. I could even deny the station from placing outbound calls if  
AGENT is blank, meaning that no agent has logged in yet.


4) Finally, I would create an extension for an agent to log out, that  
could look like this:


RemoveQueueMember(queue_name)
DBDel(agents/${CHANNEL})
Playback(agent-loggedoff)
Hangup()

where CHANNEL would be a cleaned version of the regular CHANNEL variable  
with only the technology and peer name, for example, only SIP/1234  
instead of SIP/1234-1f89


This would remove the entry in astDB at the same time it removed the  
station from the queue.


I haven't tried any of it but I think it could work.

Thanks,
Waldo

On Oct 21, 2005, at 3:39 AM, Lenz wrote:



Hello Waldo,
if you use AddQueueMember plus a fake queue_log registration, you can  
tell who the agent was, not just from what terminal she was connecting  
from. It is then possible to report who was available at a certain  
time, or see agents logging on and off, going to pause, measuring the  
average call length per agent, pause time, etc. If you run a call  
center you will want these pieces of information, otherwise you have no  
means of understanding what is going on. Simply connecting to terminals  
is not good, because you'll usually have more agents than terminals (to  
compensate for shifts, sick leave, vacations, etc)

Bye
l.


On Thu, 20 Oct 2005 17:41:31 +0200, Waldo Rubinstein  
[EMAIL PROTECTED] wrote:




I have played with AddQueueMember and it works great. However, there
is one problem that I have and I hope someone can point me in the
right direction.

My client's agents rotate seats. This means that if I want to track
calls by agent, I can't with AddQueueMember. When I look at the CDR,
it tells me the calls made/received by the station (regardless of
technology - SIP/AIX/etc). But, at any given point, I don't know
which agent made the call.

In reality, even with AgentCallBackLogin I can't tell which agent
made or received the call. Is there a way that I can identify in the
CDR which agent 

Re: [Asterisk-Users] Multitenant Call Center Setup

[Waldo Rubinstein]

I have played with AddQueueMember and it works great. However, there is one problem that I have and I hope someone can point me in the right direction.My client's agents rotate seats. This means that if I want to track calls by agent, I can't with AddQueueMember. When I look at the CDR, it tells me the calls made/received by the station (regardless of technology - SIP/AIX/etc). But, at any given point, I don't know which agent made the call.In reality, even with AgentCallBackLogin I can't tell which agent made or received the call. Is there a way that I can identify in the CDR which agent actually received or placed a call regardless of which extension he/she may be sitting on?Thanks,WaldoOn Oct 10, 2005, at 12:22 PM, Waldo Rubinstein wrote:BJ,Thanks for the prompt response. Both my clients work by using the AgentCallBackLogin so that * can send queued calls to them regardless of which SIP phone they're sitting on (sorry I didn't include this in my original email)You mean to say that if I use AddQueueMember, I could do the same and still be able to have two agents 1001?Thanks,WaldoOn Oct 10, 2005, at 11:38 AM, BJ Weschke wrote: There isn't a way to do it in agents.conf.     That being said though, there are folks that have forgone agents.conf and have used the AddQueueMember and RemoveQueueMember commands via both the dial plan and manager interfaces to work their own agents approach that certainly could be designed to support a multi-tenant environment.   On 10/10/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: Hi list (again),I have another question which I have not been able to resolve fromneither the wiki nor Google. I've been able to configure a multi-tenant setup of asterisk for 2small call centers with no problem, by simply playing with contexts(which I guess is how everyone else is doing it).The problem I have is that I've only been able to configure one global agents.conf file. This restricts my setup in a way that Icannot have two agents 1001, for example if my clients wanted to. Isthere a way to overcome this?Thanks,Waldo___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Multitenant Call Center Setup

[Waldo Rubinstein]

Hi list (again),

I have another question which I have not been able to resolve from  
neither the wiki nor Google.


I've been able to configure a multi-tenant setup of asterisk for 2  
small call centers with no problem, by simply playing with contexts  
(which I guess is how everyone else is doing it).


The problem I have is that I've only been able to configure one  
global agents.conf file. This restricts my setup in a way that I  
cannot have two agents 1001, for example if my clients wanted to. Is  
there a way to overcome this?


Thanks,
Waldo
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Multitenant Call Center Setup

[BJ Weschke]

There isn't a way to do it in agents.conf. 

That being said though, there are folks that have forgone agents.conf and have used the AddQueueMember and RemoveQueueMember commands via both the dial plan and manager interfaces to work their own agents approach that certainly could be designed to support a multi-tenant environment. 

On 10/10/05, Waldo Rubinstein [EMAIL PROTECTED] wrote:
Hi list (again),I have another question which I have not been able to resolve fromneither the wiki nor Google.
I've been able to configure a multi-tenant setup of asterisk for 2small call centers with no problem, by simply playing with contexts(which I guess is how everyone else is doing it).The problem I have is that I've only been able to configure one
global agents.conf file. This restricts my setup in a way that Icannot have two agents 1001, for example if my clients wanted to. Isthere a way to overcome this?Thanks,Waldo___
--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Multitenant Call Center Setup

[Waldo Rubinstein]

BJ,Thanks for the prompt response. Both my clients work by using the AgentCallBackLogin so that * can send queued calls to them regardless of which SIP phone they're sitting on (sorry I didn't include this in my original email)You mean to say that if I use AddQueueMember, I could do the same and still be able to have two agents 1001?Thanks,WaldoOn Oct 10, 2005, at 11:38 AM, BJ Weschke wrote: There isn't a way to do it in agents.conf.     That being said though, there are folks that have forgone agents.conf and have used the AddQueueMember and RemoveQueueMember commands via both the dial plan and manager interfaces to work their own agents approach that certainly could be designed to support a multi-tenant environment.   On 10/10/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: Hi list (again),I have another question which I have not been able to resolve fromneither the wiki nor Google. I've been able to configure a multi-tenant setup of asterisk for 2small call centers with no problem, by simply playing with contexts(which I guess is how everyone else is doing it).The problem I have is that I've only been able to configure one global agents.conf file. This restricts my setup in a way that Icannot have two agents 1001, for example if my clients wanted to. Isthere a way to overcome this?Thanks,Waldo___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Inbound call center - reliability \ scalability with queues

[Warren Smith]

I would say that we would need to be able to scale to the 200+ consecutive
call range in the near future (6 months), and hopefully to the 500+ within
the next two years.

We plan on using at least 1+0 raid with a minimum of 4 disks (minimum of 10K
rpm SCSI).  We also are planning on recording all queue calls that are
answered, and possibly outbound calls made by support agents since it has
proven to be extremely helpful when it comes to training.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Monday, May 23, 2005 6:41 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Inbound call center - reliability \
scalability with queues

For an inbound call center with 4 T1s and 30-50 agents on you would do just
fine with a single, one-processor machine. We have handled more than this on
a single P4 server although we use astGUIclient instead of Asterisk queues,
but the load is very similar. I would recommend a Sangoma Quad T1 card
because they are about 30% more efficient than Digium T1 cards. 

When you say that you need to scale to 100s of consecutive calls, is that
closer to 200 or 900? and what timeframe is that planned for?

We have a distributed in/outbound call center environment across 4
geographic locations with over 20 T1s connected so it is possible for
Asterisk to handle over 1000 consecutive calls across the system if you
design it right. One of the reasons we don't use Asterisk queues, other than
the difficulty in customizing the code to work with the ManagerAPI and
client apps, is that it was hard to scale across multiple servers. That's
why we use the astGUIclient suite which is more customizable and scalable
across multiple servers, although (and it pains me to say this because we
developed it) it is not as easy to install and setup than just creating an
Asterisk queue.

We use a combination of SIP, IAX and Zap client phones depending on the
system and the user and yes 711 is always best to use when you can. And if
you have many remote phones using a codec like GSM it may actually be better
to have a dedicated machine doing nothing but the transcoding from GSM-711
and then just using IAX or a crossover Zap T1 to the inbound server to
reduce processor load.

In any case it is always advisable to have a backup server that is fully
ready to jump in production with a minimum of reconfiguration.

A couple more questions, will you do much recording? and what kind of disks
do you plan on using?

Hope this helps,

MATT---

-Original Message-
From: Warren Smith [mailto:[EMAIL PROTECTED]
Sent: Monday, May 23, 2005 8:12 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Inbound call center - reliability \ scalability
with queues


We are wanting to move off of our legacy inter-tel phoneswitch and move to
VoIP and asterisk.  We are looking for a new PBX because the inter-tel
switch is too difficult to integrate our existing (and new) software into.  

We are a technical support center.  All our calls currently come in on toll
free numbers via T1's, and there are 3 of them.  I want to use a media
gateway to convert the T1's into SIP VOIP (I want reliable hardware for the
gateway), and use asterisk as the PBX having all incoming and outgoing
channels as SIP.  Almost all dialplans will be using Queues, and there will
likely be no more than 10 queues, with (currently) about 80 incoming
toll-free numbers.  There are approximatley 30 agents, but as of right now
there are no more than 15 agents logged in at a time.  We need to be able to
support 60-70 simultaneous calls initially and we have to be able to do this
reliably.  We also need to be able to scale into the 100's of simultanous
calls range.

What would be the best option, to have 2 powerful machines (dual
powersupply, ) with one as a hotswappable backup or have multiple machines
with a sort of load balancer setup?  Having multiple machines could possibly
cost less, but I'm not sure how the queues and agents would be managed
across multiple machines.  I.e. how would the agent 'login' to each asterisk
machine so that the calls could be handed to it, and how would the calls get
handed to an available agent by 4 seperate asterisk machines?  I've read
through the wiki, but I'm not sure how much overhead queues would put into
the system.  I want to have all the codecs the same, so the asterisk
machines doesn't do any transcoding, and have all channels as SIP.  There
will be music on hold.  Would a dual 2.8 ghz xeon in this config be able to
handle 80 simultaneous incoming calls?  Would using the 711 codec make a
difference in available processing power?

I'm sorry if this has been answered a million times already, I just didn't
see many configs close to what we're trying to do to compare to.  Thanks for
any input you may have.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http

RE: [Asterisk-Users] Inbound call center - reliability \ scalabil ity with queues

[mattf]

OK, If you are going to be recording all calls you will need to rethink
things a bit. Recording calls limits you to 50-60 consecutive conversations
per server before audio distortion starts to occur.  You will probably want
to think about limiting yourself to 3 T1s per machine. There are many ways
to set this up and I think you will probably have to go through some
trial-and-error before you find the perfect system layout for your
operations.

I would first try setting up machines that would just have the T1s on them
and take the calls in(or out) and record them. Then have those
connect(through IAX or T1 crossover) to the servers that have your queues
and phones set up on them. You will also need some really big archiving
mechanism if you want to keep those recordings around to reference in the
future. audio recordings can take up a lot of space if you need to keep them
for 3 years like we do.

SCSI RAIDs are a great solution, but sometimes(in very loaded servers) they
can hit the PCI-bus bottleneck and have issues. You may see the
ast_channel_walk_locked warning in Asterisk when this happens.

MATT---

-Original Message-
From: Warren Smith [mailto:[EMAIL PROTECTED]
Sent: Tuesday, May 24, 2005 12:38 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Inbound call center - reliability \
scalability with queues


I would say that we would need to be able to scale to the 200+ consecutive
call range in the near future (6 months), and hopefully to the 500+ within
the next two years.

We plan on using at least 1+0 raid with a minimum of 4 disks (minimum of 10K
rpm SCSI).  We also are planning on recording all queue calls that are
answered, and possibly outbound calls made by support agents since it has
proven to be extremely helpful when it comes to training.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Monday, May 23, 2005 6:41 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Inbound call center - reliability \
scalability with queues

For an inbound call center with 4 T1s and 30-50 agents on you would do just
fine with a single, one-processor machine. We have handled more than this on
a single P4 server although we use astGUIclient instead of Asterisk queues,
but the load is very similar. I would recommend a Sangoma Quad T1 card
because they are about 30% more efficient than Digium T1 cards. 

When you say that you need to scale to 100s of consecutive calls, is that
closer to 200 or 900? and what timeframe is that planned for?

We have a distributed in/outbound call center environment across 4
geographic locations with over 20 T1s connected so it is possible for
Asterisk to handle over 1000 consecutive calls across the system if you
design it right. One of the reasons we don't use Asterisk queues, other than
the difficulty in customizing the code to work with the ManagerAPI and
client apps, is that it was hard to scale across multiple servers. That's
why we use the astGUIclient suite which is more customizable and scalable
across multiple servers, although (and it pains me to say this because we
developed it) it is not as easy to install and setup than just creating an
Asterisk queue.

We use a combination of SIP, IAX and Zap client phones depending on the
system and the user and yes 711 is always best to use when you can. And if
you have many remote phones using a codec like GSM it may actually be better
to have a dedicated machine doing nothing but the transcoding from GSM-711
and then just using IAX or a crossover Zap T1 to the inbound server to
reduce processor load.

In any case it is always advisable to have a backup server that is fully
ready to jump in production with a minimum of reconfiguration.

A couple more questions, will you do much recording? and what kind of disks
do you plan on using?

Hope this helps,

MATT---

-Original Message-
From: Warren Smith [mailto:[EMAIL PROTECTED]
Sent: Monday, May 23, 2005 8:12 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Inbound call center - reliability \ scalability
with queues


We are wanting to move off of our legacy inter-tel phoneswitch and move to
VoIP and asterisk.  We are looking for a new PBX because the inter-tel
switch is too difficult to integrate our existing (and new) software into.  

We are a technical support center.  All our calls currently come in on toll
free numbers via T1's, and there are 3 of them.  I want to use a media
gateway to convert the T1's into SIP VOIP (I want reliable hardware for the
gateway), and use asterisk as the PBX having all incoming and outgoing
channels as SIP.  Almost all dialplans will be using Queues, and there will
likely be no more than 10 queues, with (currently) about 80 incoming
toll-free numbers.  There are approximatley 30 agents, but as of right now
there are no more than 15 agents logged in at a time.  We need to be able

Re: [Asterisk-Users] Inbound call center - reliability \ scalabil ity with queues

[Ilan Rabinovitch]

Matt,

Are you doing any call recording / monitoring?  What percentage?  

Ilan

On 5/23/05, mattf [EMAIL PROTECTED] wrote:
 For an inbound call center with 4 T1s and 30-50 agents on you would do just
 fine with a single, one-processor machine. We have handled more than this on
 a single P4 server although we use astGUIclient instead of Asterisk queues,
 but the load is very similar. I would recommend a Sangoma Quad T1 card
 because they are about 30% more efficient than Digium T1 cards.
 
 When you say that you need to scale to 100s of consecutive calls, is that
 closer to 200 or 900? and what timeframe is that planned for?
 
 We have a distributed in/outbound call center environment across 4
 geographic locations with over 20 T1s connected so it is possible for
 Asterisk to handle over 1000 consecutive calls across the system if you
 design it right. One of the reasons we don't use Asterisk queues, other than
 the difficulty in customizing the code to work with the ManagerAPI and
 client apps, is that it was hard to scale across multiple servers. That's
 why we use the astGUIclient suite which is more customizable and scalable
 across multiple servers, although (and it pains me to say this because we
 developed it) it is not as easy to install and setup than just creating an
 Asterisk queue.
 
 We use a combination of SIP, IAX and Zap client phones depending on the
 system and the user and yes 711 is always best to use when you can. And if
 you have many remote phones using a codec like GSM it may actually be better
 to have a dedicated machine doing nothing but the transcoding from GSM-711
 and then just using IAX or a crossover Zap T1 to the inbound server to
 reduce processor load.
 
 In any case it is always advisable to have a backup server that is fully
 ready to jump in production with a minimum of reconfiguration.
 
 A couple more questions, will you do much recording? and what kind of disks
 do you plan on using?
 
 Hope this helps,
 
 MATT---
 
 -Original Message-
 From: Warren Smith [mailto:[EMAIL PROTECTED]
 Sent: Monday, May 23, 2005 8:12 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Inbound call center - reliability \ scalability
 with queues
 
 
 We are wanting to move off of our legacy inter-tel phoneswitch and move to
 VoIP and asterisk.  We are looking for a new PBX because the inter-tel
 switch is too difficult to integrate our existing (and new) software into.
 
 We are a technical support center.  All our calls currently come in on toll
 free numbers via T1's, and there are 3 of them.  I want to use a media
 gateway to convert the T1's into SIP VOIP (I want reliable hardware for the
 gateway), and use asterisk as the PBX having all incoming and outgoing
 channels as SIP.  Almost all dialplans will be using Queues, and there will
 likely be no more than 10 queues, with (currently) about 80 incoming
 toll-free numbers.  There are approximatley 30 agents, but as of right now
 there are no more than 15 agents logged in at a time.  We need to be able to
 support 60-70 simultaneous calls initially and we have to be able to do this
 reliably.  We also need to be able to scale into the 100's of simultanous
 calls range.
 
 What would be the best option, to have 2 powerful machines (dual
 powersupply, ) with one as a hotswappable backup or have multiple machines
 with a sort of load balancer setup?  Having multiple machines could possibly
 cost less, but I'm not sure how the queues and agents would be managed
 across multiple machines.  I.e. how would the agent 'login' to each asterisk
 machine so that the calls could be handed to it, and how would the calls get
 handed to an available agent by 4 seperate asterisk machines?  I've read
 through the wiki, but I'm not sure how much overhead queues would put into
 the system.  I want to have all the codecs the same, so the asterisk
 machines doesn't do any transcoding, and have all channels as SIP.  There
 will be music on hold.  Would a dual 2.8 ghz xeon in this config be able to
 handle 80 simultaneous incoming calls?  Would using the 711 codec make a
 difference in available processing power?
 
 I'm sorry if this has been answered a million times already, I just didn't
 see many configs close to what we're trying to do to compare to.  Thanks for
 any input you may have.
 ___
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Inbound call center - reliability \ scalabil ity with queues

[mattf]

We have several different setups, but on a couple servers we are doing upto
50 concurrent conversations of recording. We ran into the 50-60 recording
ceiling about a year ago and it's mostly the hard drive that limits it to
that number, really it's a lot if you think about it, Asterisk is having the
hard drive write 100-120 audio files(-in and -out for each conversation)
several times a second. It is also important to note that we mix them with
sox after hours to reduce load on the system and load on the drives.
Although this does mean that the recordings are not available until the next
day. We also have setup 2 systems to copy the in and out files off to
another machine to be mixed more closely to realtime so that is an
alternative.

MATT---


-Original Message-
From: Ilan Rabinovitch [mailto:[EMAIL PROTECTED]
Sent: Tuesday, May 24, 2005 1:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Inbound call center - reliability \
scalabil ity with queues


Matt,

Are you doing any call recording / monitoring?  What percentage?  

Ilan

On 5/23/05, mattf [EMAIL PROTECTED] wrote:
 For an inbound call center with 4 T1s and 30-50 agents on you would do
just
 fine with a single, one-processor machine. We have handled more than this
on
 a single P4 server although we use astGUIclient instead of Asterisk
queues,
 but the load is very similar. I would recommend a Sangoma Quad T1 card
 because they are about 30% more efficient than Digium T1 cards.
 
 When you say that you need to scale to 100s of consecutive calls, is that
 closer to 200 or 900? and what timeframe is that planned for?
 
 We have a distributed in/outbound call center environment across 4
 geographic locations with over 20 T1s connected so it is possible for
 Asterisk to handle over 1000 consecutive calls across the system if you
 design it right. One of the reasons we don't use Asterisk queues, other
than
 the difficulty in customizing the code to work with the ManagerAPI and
 client apps, is that it was hard to scale across multiple servers. That's
 why we use the astGUIclient suite which is more customizable and scalable
 across multiple servers, although (and it pains me to say this because we
 developed it) it is not as easy to install and setup than just creating an
 Asterisk queue.
 
 We use a combination of SIP, IAX and Zap client phones depending on the
 system and the user and yes 711 is always best to use when you can. And if
 you have many remote phones using a codec like GSM it may actually be
better
 to have a dedicated machine doing nothing but the transcoding from
GSM-711
 and then just using IAX or a crossover Zap T1 to the inbound server to
 reduce processor load.
 
 In any case it is always advisable to have a backup server that is fully
 ready to jump in production with a minimum of reconfiguration.
 
 A couple more questions, will you do much recording? and what kind of
disks
 do you plan on using?
 
 Hope this helps,
 
 MATT---
 
 -Original Message-
 From: Warren Smith [mailto:[EMAIL PROTECTED]
 Sent: Monday, May 23, 2005 8:12 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Inbound call center - reliability \ scalability
 with queues
 
 
 We are wanting to move off of our legacy inter-tel phoneswitch and move to
 VoIP and asterisk.  We are looking for a new PBX because the inter-tel
 switch is too difficult to integrate our existing (and new) software into.
 
 We are a technical support center.  All our calls currently come in on
toll
 free numbers via T1's, and there are 3 of them.  I want to use a media
 gateway to convert the T1's into SIP VOIP (I want reliable hardware for
the
 gateway), and use asterisk as the PBX having all incoming and outgoing
 channels as SIP.  Almost all dialplans will be using Queues, and there
will
 likely be no more than 10 queues, with (currently) about 80 incoming
 toll-free numbers.  There are approximatley 30 agents, but as of right now
 there are no more than 15 agents logged in at a time.  We need to be able
to
 support 60-70 simultaneous calls initially and we have to be able to do
this
 reliably.  We also need to be able to scale into the 100's of simultanous
 calls range.
 
 What would be the best option, to have 2 powerful machines (dual
 powersupply, ) with one as a hotswappable backup or have multiple machines
 with a sort of load balancer setup?  Having multiple machines could
possibly
 cost less, but I'm not sure how the queues and agents would be managed
 across multiple machines.  I.e. how would the agent 'login' to each
asterisk
 machine so that the calls could be handed to it, and how would the calls
get
 handed to an available agent by 4 seperate asterisk machines?  I've read
 through the wiki, but I'm not sure how much overhead queues would put into
 the system.  I want to have all the codecs the same, so the asterisk
 machines doesn't do any transcoding, and have all channels as SIP

RE: [Asterisk-Users] Inbound call center - reliability \ scalability with queues

[Warren Smith]

The asterisk machines will not have anything to do with the T1's, when they
receive the call it will be SIP VOIP.  There will be media gateways (i.e.
cisco media gateways) to change all T1 signals to VOIP before it reaches the
PBX.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Tuesday, May 24, 2005 11:42 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Inbound call center - reliability \
scalability with queues

OK, If you are going to be recording all calls you will need to rethink
things a bit. Recording calls limits you to 50-60 consecutive conversations
per server before audio distortion starts to occur.  You will probably want
to think about limiting yourself to 3 T1s per machine. There are many ways
to set this up and I think you will probably have to go through some
trial-and-error before you find the perfect system layout for your
operations.

I would first try setting up machines that would just have the T1s on them
and take the calls in(or out) and record them. Then have those
connect(through IAX or T1 crossover) to the servers that have your queues
and phones set up on them. You will also need some really big archiving
mechanism if you want to keep those recordings around to reference in the
future. audio recordings can take up a lot of space if you need to keep them
for 3 years like we do.

SCSI RAIDs are a great solution, but sometimes(in very loaded servers) they
can hit the PCI-bus bottleneck and have issues. You may see the
ast_channel_walk_locked warning in Asterisk when this happens.

MATT---

-Original Message-
From: Warren Smith [mailto:[EMAIL PROTECTED]
Sent: Tuesday, May 24, 2005 12:38 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Inbound call center - reliability \
scalability with queues


I would say that we would need to be able to scale to the 200+ consecutive
call range in the near future (6 months), and hopefully to the 500+ within
the next two years.

We plan on using at least 1+0 raid with a minimum of 4 disks (minimum of 10K
rpm SCSI).  We also are planning on recording all queue calls that are
answered, and possibly outbound calls made by support agents since it has
proven to be extremely helpful when it comes to training.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Monday, May 23, 2005 6:41 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Inbound call center - reliability \
scalability with queues

For an inbound call center with 4 T1s and 30-50 agents on you would do just
fine with a single, one-processor machine. We have handled more than this on
a single P4 server although we use astGUIclient instead of Asterisk queues,
but the load is very similar. I would recommend a Sangoma Quad T1 card
because they are about 30% more efficient than Digium T1 cards. 

When you say that you need to scale to 100s of consecutive calls, is that
closer to 200 or 900? and what timeframe is that planned for?

We have a distributed in/outbound call center environment across 4
geographic locations with over 20 T1s connected so it is possible for
Asterisk to handle over 1000 consecutive calls across the system if you
design it right. One of the reasons we don't use Asterisk queues, other than
the difficulty in customizing the code to work with the ManagerAPI and
client apps, is that it was hard to scale across multiple servers. That's
why we use the astGUIclient suite which is more customizable and scalable
across multiple servers, although (and it pains me to say this because we
developed it) it is not as easy to install and setup than just creating an
Asterisk queue.

We use a combination of SIP, IAX and Zap client phones depending on the
system and the user and yes 711 is always best to use when you can. And if
you have many remote phones using a codec like GSM it may actually be better
to have a dedicated machine doing nothing but the transcoding from GSM-711
and then just using IAX or a crossover Zap T1 to the inbound server to
reduce processor load.

In any case it is always advisable to have a backup server that is fully
ready to jump in production with a minimum of reconfiguration.

A couple more questions, will you do much recording? and what kind of disks
do you plan on using?

Hope this helps,

MATT---

-Original Message-
From: Warren Smith [mailto:[EMAIL PROTECTED]
Sent: Monday, May 23, 2005 8:12 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Inbound call center - reliability \ scalability
with queues


We are wanting to move off of our legacy inter-tel phoneswitch and move to
VoIP and asterisk.  We are looking for a new PBX because the inter-tel
switch is too difficult to integrate our existing (and new) software into.  

We are a technical support center.  All our calls currently come

RE: [Asterisk-Users] Inbound call center - reliability \ scalabil ity with queues

[mattf]

Well, that really changes things then. I'm not really sure what to tell you
because we've never done it that way. The ciscos are limited in how you can
have them send calls to different servers based upon specific parameters so
you will be limited there somewhat. Is there a specific reason you're not
going to use Asterisk servers for the T1-SIP conversion? 

It would allow you to do the recording up front and give you more control
over call handling, and I can't imagine that you can get a new 3 x T1 Cisco
VOIP machine for less than $5000 like you can with Asterisk.

MATT---

-Original Message-
From: Warren Smith [mailto:[EMAIL PROTECTED]
Sent: Tuesday, May 24, 2005 2:43 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Inbound call center - reliability \
scalability with queues


The asterisk machines will not have anything to do with the T1's, when they
receive the call it will be SIP VOIP.  There will be media gateways (i.e.
cisco media gateways) to change all T1 signals to VOIP before it reaches the
PBX.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Tuesday, May 24, 2005 11:42 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Inbound call center - reliability \
scalability with queues

OK, If you are going to be recording all calls you will need to rethink
things a bit. Recording calls limits you to 50-60 consecutive conversations
per server before audio distortion starts to occur.  You will probably want
to think about limiting yourself to 3 T1s per machine. There are many ways
to set this up and I think you will probably have to go through some
trial-and-error before you find the perfect system layout for your
operations.

I would first try setting up machines that would just have the T1s on them
and take the calls in(or out) and record them. Then have those
connect(through IAX or T1 crossover) to the servers that have your queues
and phones set up on them. You will also need some really big archiving
mechanism if you want to keep those recordings around to reference in the
future. audio recordings can take up a lot of space if you need to keep them
for 3 years like we do.

SCSI RAIDs are a great solution, but sometimes(in very loaded servers) they
can hit the PCI-bus bottleneck and have issues. You may see the
ast_channel_walk_locked warning in Asterisk when this happens.

MATT---

-Original Message-
From: Warren Smith [mailto:[EMAIL PROTECTED]
Sent: Tuesday, May 24, 2005 12:38 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Inbound call center - reliability \
scalability with queues


I would say that we would need to be able to scale to the 200+ consecutive
call range in the near future (6 months), and hopefully to the 500+ within
the next two years.

We plan on using at least 1+0 raid with a minimum of 4 disks (minimum of 10K
rpm SCSI).  We also are planning on recording all queue calls that are
answered, and possibly outbound calls made by support agents since it has
proven to be extremely helpful when it comes to training.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Monday, May 23, 2005 6:41 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Inbound call center - reliability \
scalability with queues

For an inbound call center with 4 T1s and 30-50 agents on you would do just
fine with a single, one-processor machine. We have handled more than this on
a single P4 server although we use astGUIclient instead of Asterisk queues,
but the load is very similar. I would recommend a Sangoma Quad T1 card
because they are about 30% more efficient than Digium T1 cards. 

When you say that you need to scale to 100s of consecutive calls, is that
closer to 200 or 900? and what timeframe is that planned for?

We have a distributed in/outbound call center environment across 4
geographic locations with over 20 T1s connected so it is possible for
Asterisk to handle over 1000 consecutive calls across the system if you
design it right. One of the reasons we don't use Asterisk queues, other than
the difficulty in customizing the code to work with the ManagerAPI and
client apps, is that it was hard to scale across multiple servers. That's
why we use the astGUIclient suite which is more customizable and scalable
across multiple servers, although (and it pains me to say this because we
developed it) it is not as easy to install and setup than just creating an
Asterisk queue.

We use a combination of SIP, IAX and Zap client phones depending on the
system and the user and yes 711 is always best to use when you can. And if
you have many remote phones using a codec like GSM it may actually be better
to have a dedicated machine doing nothing but the transcoding from GSM-711
and then just using IAX or a crossover Zap

[Asterisk-Users] Inbound call center - reliability \ scalability with queues

[Warren Smith]

We are wanting to 
move off of our legacy inter-tel phoneswitch and move to VoIP and 
asterisk. We are looking for a new PBX because the inter-tel switch is too 
difficult to integrate our existing (and new) software into. 


We are a technical 
support center. All our calls currently come in on toll free numbers via 
T1's, and there are 3 of them. I want to use a media gateway to convert 
the T1's into SIP VOIP (I want reliable hardware for the gateway), and use 
asterisk as the PBX having all incoming and outgoing channels as SIP. 
Almost all dialplans will be using Queues, and there will likely be no more than 
10 queues, with (currently) about 80 incoming toll-free numbers. There are 
approximatley 30 agents, but as of right now there are no more than 15 agents 
logged in at a time. We need to be able to support 60-70 simultaneous 
calls initially and we have to be able to do this reliably. We also need 
to be able to scale into the 100's of simultanous calls 
range.

What would be the 
best option, to have 2 powerful machines (dual powersupply, ) with one as a 
hotswappable backup or have multiple machines with a sort of load balancer 
setup? Having multiple machines could possibly cost less, but I'm not sure 
how the queues and agents would be managed across multiple machines. I.e. 
how would the agent 'login' to each asterisk machine so that the calls could be 
handed to it, and how would the calls get handed to an available agent by 4 
seperate asterisk machines? I've read through the wiki, but I'm not sure 
how much overhead queues would put into the system. I want to have all the 
codecs the same, so the asterisk machines doesn't do any transcoding, and have 
all channels as SIP. There will be music on hold. Would a dual 2.8 
ghz xeon in this config be able to handle 80 simultaneous incoming calls? 
Would using the 711 codec make a difference in available processing 
power?

I'm sorry if this 
has been answered a million times already, I just didn't see many configs close 
to what we're trying to do to compare to. Thanks for any input you may 
have.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Inbound call center - reliability \ scalabil ity with queues

[mattf]

For an inbound call center with 4 T1s and 30-50 agents on you would do just
fine with a single, one-processor machine. We have handled more than this on
a single P4 server although we use astGUIclient instead of Asterisk queues,
but the load is very similar. I would recommend a Sangoma Quad T1 card
because they are about 30% more efficient than Digium T1 cards. 

When you say that you need to scale to 100s of consecutive calls, is that
closer to 200 or 900? and what timeframe is that planned for?

We have a distributed in/outbound call center environment across 4
geographic locations with over 20 T1s connected so it is possible for
Asterisk to handle over 1000 consecutive calls across the system if you
design it right. One of the reasons we don't use Asterisk queues, other than
the difficulty in customizing the code to work with the ManagerAPI and
client apps, is that it was hard to scale across multiple servers. That's
why we use the astGUIclient suite which is more customizable and scalable
across multiple servers, although (and it pains me to say this because we
developed it) it is not as easy to install and setup than just creating an
Asterisk queue.

We use a combination of SIP, IAX and Zap client phones depending on the
system and the user and yes 711 is always best to use when you can. And if
you have many remote phones using a codec like GSM it may actually be better
to have a dedicated machine doing nothing but the transcoding from GSM-711
and then just using IAX or a crossover Zap T1 to the inbound server to
reduce processor load.

In any case it is always advisable to have a backup server that is fully
ready to jump in production with a minimum of reconfiguration.

A couple more questions, will you do much recording? and what kind of disks
do you plan on using?

Hope this helps,

MATT---

-Original Message-
From: Warren Smith [mailto:[EMAIL PROTECTED]
Sent: Monday, May 23, 2005 8:12 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Inbound call center - reliability \ scalability
with queues


We are wanting to move off of our legacy inter-tel phoneswitch and move to
VoIP and asterisk.  We are looking for a new PBX because the inter-tel
switch is too difficult to integrate our existing (and new) software into.  

We are a technical support center.  All our calls currently come in on toll
free numbers via T1's, and there are 3 of them.  I want to use a media
gateway to convert the T1's into SIP VOIP (I want reliable hardware for the
gateway), and use asterisk as the PBX having all incoming and outgoing
channels as SIP.  Almost all dialplans will be using Queues, and there will
likely be no more than 10 queues, with (currently) about 80 incoming
toll-free numbers.  There are approximatley 30 agents, but as of right now
there are no more than 15 agents logged in at a time.  We need to be able to
support 60-70 simultaneous calls initially and we have to be able to do this
reliably.  We also need to be able to scale into the 100's of simultanous
calls range.

What would be the best option, to have 2 powerful machines (dual
powersupply, ) with one as a hotswappable backup or have multiple machines
with a sort of load balancer setup?  Having multiple machines could possibly
cost less, but I'm not sure how the queues and agents would be managed
across multiple machines.  I.e. how would the agent 'login' to each asterisk
machine so that the calls could be handed to it, and how would the calls get
handed to an available agent by 4 seperate asterisk machines?  I've read
through the wiki, but I'm not sure how much overhead queues would put into
the system.  I want to have all the codecs the same, so the asterisk
machines doesn't do any transcoding, and have all channels as SIP.  There
will be music on hold.  Would a dual 2.8 ghz xeon in this config be able to
handle 80 simultaneous incoming calls?  Would using the 711 codec make a
difference in available processing power?

I'm sorry if this has been answered a million times already, I just didn't
see many configs close to what we're trying to do to compare to.  Thanks for
any input you may have.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Simple Call Center Setup?

[Zarjazz]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

I am looking to implement a simple call center with Asterisk and I'm
scratching my head trying to work out the best setup.

Now the usual dialin / dialout setup I have no problems with. SIP phone
on each desk, * acting as the SIP proxy / PSTN gateway. User dials into
the gateway to talk to an agent (hunt group type thing), or a single
agent dials out to a specific number. That part is simple enough and
works already.

However I need to add the following feature:

Some application (asterisk maybe or a custom SIP softphone type thing)
needs to dial out to a range of numbers, once a success connection is
made I need to then transfer the call to the first free agent. The
available agents need to notify the system somehow that they are ready.
I was thinking of maybe 'parking' and then somehow inviting the remote
call to their phone.

So is something like this possible already with * or do I need to go and
start getting out my C refence manuals and do some coding?

Vince.
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (MingW32)

iD8DBQFAN1CaPx/nyuA99rgRAolvAJ4ir83H4UlNeYW4twPG+w0j5KRBgwCg1pXh
2IDBbj/+dUh/w5E1BmH4kDM=
=NX9Q
-END PGP SIGNATURE-
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Simple Call Center Setup?

[Matthew B Marlowe]

This was just discussed yesterday I believe, because I read it.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zarjazz
Sent: Saturday, February 21, 2004 7:36 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Simple Call Center Setup?

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

I am looking to implement a simple call center with Asterisk and I'm
scratching my head trying to work out the best setup.

Now the usual dialin / dialout setup I have no problems with. SIP phone
on each desk, * acting as the SIP proxy / PSTN gateway. User dials into
the gateway to talk to an agent (hunt group type thing), or a single
agent dials out to a specific number. That part is simple enough and
works already.

However I need to add the following feature:

Some application (asterisk maybe or a custom SIP softphone type thing)
needs to dial out to a range of numbers, once a success connection is
made I need to then transfer the call to the first free agent. The
available agents need to notify the system somehow that they are ready.
I was thinking of maybe 'parking' and then somehow inviting the remote
call to their phone.

So is something like this possible already with * or do I need to go and
start getting out my C refence manuals and do some coding?

Vince.
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (MingW32)

iD8DBQFAN1CaPx/nyuA99rgRAolvAJ4ir83H4UlNeYW4twPG+w0j5KRBgwCg1pXh
2IDBbj/+dUh/w5E1BmH4kDM=
=NX9Q
-END PGP SIGNATURE-
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Simple Call Center Setup?

[Zarjazz]

Which thread exactly? There is the agents / ackcall one but that is for
*inbound* calls which I already can do. This is for a dialout redirect
type scenario.

Vince.


Matthew B Marlowe wrote:

 This was just discussed yesterday I believe, because I read it.

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Zarjazz
 Sent: Saturday, February 21, 2004 7:36 AM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Simple Call Center Setup?

 I am looking to implement a simple call center with Asterisk and I'm
 scratching my head trying to work out the best setup.
 
 Now the usual dialin / dialout setup I have no problems with. SIP phone
 on each desk, * acting as the SIP proxy / PSTN gateway. User dials into
 the gateway to talk to an agent (hunt group type thing), or a single
 agent dials out to a specific number. That part is simple enough and
 works already.
 
 However I need to add the following feature:
 
 Some application (asterisk maybe or a custom SIP softphone type thing)
 needs to dial out to a range of numbers, once a success connection is
 made I need to then transfer the call to the first free agent. The
 available agents need to notify the system somehow that they are ready.
 I was thinking of maybe 'parking' and then somehow inviting the remote
 call to their phone.
 
 So is something like this possible already with * or do I need to go and
 start getting out my C refence manuals and do some coding?
 
 Vince.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Simple Call Center Setup?

[TC]

Come meet me in #asterisk-dev (tclark)
if you have pressing need  have 'C' skill's we have just about got a alpha
of this, 
This is a new * app we are calling ICD
Intelligent Call Distributor :)


- Original Message - 
From: Zarjazz [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, February 21, 2004 4:35 AM
Subject: [Asterisk-Users] Simple Call Center Setup?


 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1
 
 I am looking to implement a simple call center with Asterisk and I'm
 scratching my head trying to work out the best setup.
 
 Now the usual dialin / dialout setup I have no problems with. SIP phone
 on each desk, * acting as the SIP proxy / PSTN gateway. User dials into
 the gateway to talk to an agent (hunt group type thing), or a single
 agent dials out to a specific number. That part is simple enough and
 works already.
 
 However I need to add the following feature:
 
 Some application (asterisk maybe or a custom SIP softphone type thing)
 needs to dial out to a range of numbers, once a success connection is
 made I need to then transfer the call to the first free agent. The
 available agents need to notify the system somehow that they are ready.
 I was thinking of maybe 'parking' and then somehow inviting the remote
 call to their phone.
 
 So is something like this possible already with * or do I need to go and
 start getting out my C refence manuals and do some coding?
 
 Vince.
 -BEGIN PGP SIGNATURE-
 Version: GnuPG v1.2.4 (MingW32)
 
 iD8DBQFAN1CaPx/nyuA99rgRAolvAJ4ir83H4UlNeYW4twPG+w0j5KRBgwCg1pXh
 2IDBbj/+dUh/w5E1BmH4kDM=
 =NX9Q
 -END PGP SIGNATURE-
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] * For Call Center

[Steve]

On Thursday 15 January 2004 12:03 am, [EMAIL PROTECTED] wrote:
 sounds like one of those pesky auto dialers the simpsons make fun of.

It sure does...


-- 
Steve

__
You actually need to constantly be alert 
 and willing to handle things, or life 
   will find a way to get you good!
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] * For Call Center

[mattf]

Hello,

I think you need to do a little more looking around on the Asterisk
resources and on Google. What you are trying to do is mostly possible if you
have the time, patience and money to follow through with it. 

One thing you need to learn is that a great many on this list despise
telemarketers of any kind, and even more so they hate it when people don't
do their own research for questions that have been answered many times
before.

Asterisk cannot easily be turned into a predictive dialer(which is a good
thing) there are a few people that have created simple auto-dialer programs
that simply dial one number after another one-at-a-time(Myself included)
which is not very efficient for bulk cold-calling, but works wonderfully to
call back your customers on a quarterly basis for customer checkups like my
company does with it's customers.

There is a group of Asterisk users that decided to modify the code of
Asterisk to try to make it a predictive dialer, called shady_dial I believe,
but I haven't heard anything about it lately.

Back to your questions: 
- yes, Asterisk can be used for a fractional voice/data T1. there are
several people that have done this.
- you will need a decent powered x86 computer with at least a digium single
T1 card 
- yes Asterisk can be used as an auto-dialer, if you program it to do so
yourself. The easiest way is to use the manager interface or generate .call
files, but it is not very fast and will not piss people off at a high rate
like a the predictive dialer you want will
- maybe Asterisk can be turned into a predictive dialer, but you'd have to
do that yourself or find out if shady_dial has succeeded in their project
- yes you can use a screen popup system on win32 or Unix there are resources
out there to do it and it's not terribly hard to do if you understand the
APIs involved.
- no, Asterisk will not work out of the box for what you want to do, it will
take effort and time to get it all working and you may end up spending as
much as you would on a predictive dialer to get it all done.
- don't expect much sympathy from the people here on this list as the the
cost of a predictive dialer.

Thank you and good luck,

MATT---

-Original Message-
From: empire underground [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 14, 2004 11:08 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] * For Call Center


Hi Everyone ;)


   I have posted something like this before but yeilded no solid help as of 
yet.
I am new to * and havent even setup a box for it yet as to I have no clue 
what I should go ahead and buy before wasting a few $k. Im looking to setup 
* for my office with outbound calling only with some call agents, and also 
remote agents so they can work from home. At this time im not looking to do 
Voip at all... but that will change in the future. I have a T1 with 12 
anolog lines and the rest for data (768k). I need to know what cards I 
should buy? I would also like to setup the box with 12-16 lines for outbound

calling, and im nto trying to do (IVR). What I would like to do is make * 
either a predictive/auto dialer only. I read about a few people doing this 
when searching google but cant find the links anymore :( Aslo someone made a

win32 program to log into * and get screen pops of all the info that was 
dialed for that # such as address, name, phone, ect... I dont realy care if 
I have to write an agi for it in linux because I hate winblows and would 
rather stay far far away from it ;) If anyone can help or point me in the 
right direction it would be much help ;)
Also I have checked wiki allready... I cant really find anything there for 
this. Also is it even possible for this? I know I would have to write a agi 
for the screen pops to popup in web browser and rout that info to the person

logged in and waiting in that queue, I was thinking about using sql backend 
for the db and maybe writing agi to import the .csv file? Also I was 
thinking about flying someone down here to Florida if all else fails (unless

you already live here) to maybe help setup this type of box, or even giving 
root access to the box and configuring it? because a commercial  dialer 
costs WAY too much! they want anywhere from $3500-30,000 for dialers... and 
then even pay another $1,500 for a license per agent that wil be using it! 
talk about getting raped!
thanx for all you help in advance
chris

_
Scope out the new MSN Plus Internet Software - optimizes dial-up to the max!

   http://join.msn.com/?pgmarket=en-uspage=byoa/plusST=1

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo

Re: [Asterisk-Users] * For Call Center

[C. Maj]

On Thu, 15 Jan 2004, Steve waxed:

 On Thursday 15 January 2004 12:03 am, [EMAIL PROTECTED] wrote:
  sounds like one of those pesky auto dialers the simpsons make fun of.
 
 It sure does...

The AT-5000 was Prof. Frink's first patent, and it was
designed to alert children of snow days and such.  I think
Homer bought it at one of those pesky police auctions, you
know, the ones where the liberty and freedom loving US
government says your property is guilty of a crime and
theirs to sell...

But don't forget that Prof. Frink went on to invent such
wonders as the Flying Motorcycle, a Matter Transporter, and
the Frinkahedron:

http://www.internerd.com/frink.retired/frinkv.3/inventions/

--Chris


-- 

Chris Maj cmaj_hat_freedomcorpse_hot_info
Pronunciation Guide:  Maj == May
Fingerprint: 43D6 799C F6CF F920 6623  DC85 C8A3 CFFE F0DE C146

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] * For Call Center

[C. Maj]

On Thu, 15 Jan 2004, mattf waxed:

8's

 There is a group of Asterisk users that decided to modify the code of
 Asterisk to try to make it a predictive dialer, called shady_dial I believe,
 but I haven't heard anything about it lately.

http://shadydial.sourceforge.net/

Lots of recent updates made in CVS, and it works with the
latest and greatest * CVS, too.  No screen pops yet, but
that is the next step.  Call results are simply logged in
the phone, which is pretty sloppy since it resides in the
agent hangup function.

Francois Lambert posted some time ago on -dev that his
company had worked on a predictive dialer with answering
machine detection.  Said they hacked * code a little, too,
and since it's GPL I would be interested in seeing it.

--Chris


-- 

Chris Maj cmaj_hat_freedomcorpse_hot_info
Pronunciation Guide:  Maj == May
Fingerprint: 43D6 799C F6CF F920 6623  DC85 C8A3 CFFE F0DE C146

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] * For Call Center

[Sean Cheesman]

Actually he found it in the dumpster after the police threw it out
following a bust!  Does anyone want to send a dollar to Mr. Happy?!

-Original Message-
From: C. Maj [mailto:[EMAIL PROTECTED] 
Sent: Thursday, January 15, 2004 12:40 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] * For Call Center


On Thu, 15 Jan 2004, Steve waxed:

 On Thursday 15 January 2004 12:03 am, [EMAIL PROTECTED] wrote:
  sounds like one of those pesky auto dialers the simpsons make fun 
  of.
 
 It sure does...

The AT-5000 was Prof. Frink's first patent, and it was designed to
alert children of snow days and such.  I think Homer bought it at one
of those pesky police auctions, you know, the ones where the liberty and
freedom loving US government says your property is guilty of a crime and
theirs to sell...

But don't forget that Prof. Frink went on to invent such wonders as the
Flying Motorcycle, a Matter Transporter, and the Frinkahedron:

http://www.internerd.com/frink.retired/frinkv.3/inventions/

--Chris


-- 

Chris Maj cmaj_hat_freedomcorpse_hot_info
Pronunciation Guide:  Maj == May
Fingerprint: 43D6 799C F6CF F920 6623  DC85 C8A3 CFFE F0DE C146

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] * For Call Center

[empire underground]

Hi Everyone ;)

  I have posted something like this before but yeilded no solid help as of 
yet.
I am new to * and havent even setup a box for it yet as to I have no clue 
what I should go ahead and buy before wasting a few $k. Im looking to setup 
* for my office with outbound calling only with some call agents, and also 
remote agents so they can work from home. At this time im not looking to do 
Voip at all... but that will change in the future. I have a T1 with 12 
anolog lines and the rest for data (768k). I need to know what cards I 
should buy? I would also like to setup the box with 12-16 lines for outbound 
calling, and im nto trying to do (IVR). What I would like to do is make * 
either a predictive/auto dialer only. I read about a few people doing this 
when searching google but cant find the links anymore :( Aslo someone made a 
win32 program to log into * and get screen pops of all the info that was 
dialed for that # such as address, name, phone, ect... I dont realy care if 
I have to write an agi for it in linux because I hate winblows and would 
rather stay far far away from it ;) If anyone can help or point me in the 
right direction it would be much help ;)
Also I have checked wiki allready... I cant really find anything there for 
this. Also is it even possible for this? I know I would have to write a agi 
for the screen pops to popup in web browser and rout that info to the person 
logged in and waiting in that queue, I was thinking about using sql backend 
for the db and maybe writing agi to import the .csv file? Also I was 
thinking about flying someone down here to Florida if all else fails (unless 
you already live here) to maybe help setup this type of box, or even giving 
root access to the box and configuring it? because a commercial  dialer 
costs WAY too much! they want anywhere from $3500-30,000 for dialers... and 
then even pay another $1,500 for a license per agent that wil be using it! 
talk about getting raped!
thanx for all you help in advance
chris

_
Scope out the new MSN Plus Internet Software — optimizes dial-up to the max! 
  http://join.msn.com/?pgmarket=en-uspage=byoa/plusST=1

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] * For Call Center

[nanog]

sounds like one of those pesky auto dialers the simpsons make fun of.


- Original Message - 
From: empire underground [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, January 14, 2004 11:08 PM
Subject: [Asterisk-Users] * For Call Center


 Hi Everyone ;)


I have posted something like this before but yeilded no solid help as
of
 yet.
 I am new to * and havent even setup a box for it yet as to I have no clue
 what I should go ahead and buy before wasting a few $k. Im looking to
setup
 * for my office with outbound calling only with some call agents, and also
 remote agents so they can work from home. At this time im not looking to
do
 Voip at all... but that will change in the future. I have a T1 with 12
 anolog lines and the rest for data (768k). I need to know what cards I
 should buy? I would also like to setup the box with 12-16 lines for
outbound
 calling, and im nto trying to do (IVR). What I would like to do is make *
 either a predictive/auto dialer only. I read about a few people doing this
 when searching google but cant find the links anymore :( Aslo someone made
a
 win32 program to log into * and get screen pops of all the info that was
 dialed for that # such as address, name, phone, ect... I dont realy care
if
 I have to write an agi for it in linux because I hate winblows and would
 rather stay far far away from it ;) If anyone can help or point me in the
 right direction it would be much help ;)
 Also I have checked wiki allready... I cant really find anything there for
 this. Also is it even possible for this? I know I would have to write a
agi
 for the screen pops to popup in web browser and rout that info to the
person
 logged in and waiting in that queue, I was thinking about using sql
backend
 for the db and maybe writing agi to import the .csv file? Also I was
 thinking about flying someone down here to Florida if all else fails
(unless
 you already live here) to maybe help setup this type of box, or even
giving
 root access to the box and configuring it? because a commercial  dialer
 costs WAY too much! they want anywhere from $3500-30,000 for dialers...
and
 then even pay another $1,500 for a license per agent that wil be using it!
 talk about getting raped!
 thanx for all you help in advance
 chris

 _
 Scope out the new MSN Plus Internet Software - optimizes dial-up to the
max!
http://join.msn.com/?pgmarket=en-uspage=byoa/plusST=1

 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] re: call center design question

[Ben]

 Rich Adamson a écrit :
 
 Would like to deploy * in a small help desk environment (five to ten
 people) using call queues and some sort of CTI interface to pop Remedy
 screen data in front of the help desk person receiving the call. Data
 to be popped would be based on CallerID.
 
 Anyone doing something similar?
 
 Anyone interfacing to an external Remedy system?
 
 Any reference sites that I could read/learn more of the requirements
 and/or 10,000 foot implementation?
 
 Rich
 
 
 
 
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
   
 
 I deployed a small call center using Gnophone as the screen data, 
 together with dial + URL. Basically when the operator answers someone 
 from the queue, an URL is pushed and displayed in Gnophone; this is 
 quite simple as it is only web technology. The limitation is that no 
 data is displayed until the called is transfered.


I would really like to have more info about this!
Is it possible?
BTW Gnophone uses IAX. Does anybody knows if there is a good IAX softphone for 
Windows?

Ben
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users