ViciDial Queries

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ViciDial Queries

Postby coolbuddy1981 » Sun Jan 03, 2010 7:32 pm

Hi,

I m working in a call center and currently we are using Avaya predictive dialer. However my managing Director has has me to enquire about Vicidial.

We have a server [Intel xeon with 4gb of ram] is it sufficient to manage 40 seats with call recording facility.

I know how to install Centos OS 5 and i wanted to know is there any extra cards or circuits required to install Vicidial.

Currently we buy minutes from Novatel Networks. And we have a sip account with IP authentication but no username and passwords.

Thanks in advance.
Last edited by coolbuddy1981 on Sun Jan 03, 2010 7:56 pm, edited 1 time in total.
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Postby mflorell » Sun Jan 03, 2010 7:49 pm

You need to give us more information:

- what kind of trunks?
- what kind of phones?
- lines to agent ratio?
- inbound/outbound/blended call handling?
- what kind of Xeon processor(how many cores)?
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Postby coolbuddy1981 » Sun Jan 03, 2010 7:59 pm

mflorell wrote:You need to give us more information:

- what kind of trunks?
- what kind of phones?
- lines to agent ratio?
- inbound/outbound/blended call handling?
- what kind of Xeon processor(how many cores)?


I don't know anything about trunks.
-SIP phones
agent ration 3:1
outbout call handling
quadcore xeon processor.
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Postby mflorell » Mon Jan 04, 2010 1:56 am

trunks are the kind of telco lines you use to dial out on. Please list your bandwidth connection and what codec you are using if you are using SIP.

with 40 agents at 3:1 outbound dialing we usually recommend 3 servers although we have clients that run more agents on less hardware.

You don't need any special hardware to run ViciDial, depending on the kind of trunks you use.
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Postby Op3r » Mon Jan 04, 2010 12:31 pm

Hello,

I know a bit about novatel as I also use them on some of my installs.

They are only doing SIP IP authentication. Which is not hard. They also do 300 redirects.


As for 40 seats, Matt already gave you a figure which is 3 servers. Which is cool and stable depending on what hardware you are going to use.
Get paid for US outbound Toll Free calls. PM me.
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Postby coolbuddy1981 » Mon Jan 04, 2010 6:33 pm

mflorell wrote:trunks are the kind of telco lines you use to dial out on. Please list your bandwidth connection and what codec you are using if you are using SIP.

with 40 agents at 3:1 outbound dialing we usually recommend 3 servers although we have clients that run more agents on less hardware.

You don't need any special hardware to run ViciDial, depending on the kind of trunks you use.


So that means i have to purchase trunks to use with Vicidial. We have a leased line of 2MBPs and currently our provider is providing us g729 codec.
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Postby mflorell » Mon Jan 04, 2010 7:00 pm

If you are using G729 you need to buy licenses or a hardware transcoder.
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Postby Michael_N » Mon Jan 04, 2010 7:24 pm

No you don't need to buy a trunk you could use your novatel siptrunk.

But you need to buy g729 licences
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Postby coolbuddy1981 » Tue Jan 05, 2010 1:39 pm

Michael_N wrote:No you don't need to buy a trunk you could use your novatel siptrunk.

But you need to buy g729 licences


How many licenses i have to purchase for 3:1 dialing and what is the cost per license.

Is is monthly, yearly or lifetime .
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Postby mflorell » Tue Jan 05, 2010 3:12 pm

40 seats x 3 = 120 licenses

licenses are about $10 each either from Digium or Howler, so you are looking at $1200
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thanks

Postby brett05 » Tue Jan 05, 2010 5:32 pm

i use the free codec intel before many month and i have try it with more seats and no probleme seen before
www.asterisk.hosting.lv
just choose the version of asterisk we recomend asterisk 1.4.21.2 .
and choose your cpu for that and gcc4 not icc
exemple asterisk 1.4 in core2duo or dual core with os 32 bit choose this:
codec_g729-ast14-gcc4-glibc-pentium4.so
or
codec_g729-ast14-gcc4-glibc-core2.so
for asterisk 1.4 in core2duo or dual core with os 64 bit choose this:
codec_g729-ast14-gcc4-glibc-x86_64-pentium4.so
or
codec_g729-ast14-gcc4-glibc-x86_64-core2.so
etc...
put it in /usr/lib/asterisk/modules then rename it to codec_g729.so finally in cli asterisk do restart now
then open again asterisk with asterisk -rvvv after verify if your codec is updated good and do show translation
ooopsss
enjoy
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Postby mflorell » Tue Jan 05, 2010 7:10 pm

We do not approve of the use of the free G729 codecs for commercial purposes. Doing so is illegal and may result in the owner of the G729 algorithm seeking legal action against you.
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Postby Michael_N » Tue Jan 05, 2010 8:20 pm

Is a option using GSM codec?
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thanks

Postby brett05 » Tue Jan 05, 2010 8:35 pm

yes
but also i was buy before 3 licence codec from diguim there no différence the only difference is the codec from there work only in asterisk 1.4 and 1.6
but in production they are no différence i have never see a probleme
thanks
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thanks

Postby brett05 » Tue Jan 05, 2010 8:37 pm

GSM is good tools also i recomend using codec GSM then send it as quality to all agent.
but in quality he is not very good in sound but he save more our bandwith .
codec G729 buffer more the cpu and need have more processor.
but in quality he is good.
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Postby mflorell » Tue Jan 05, 2010 9:15 pm

In addition to using less CPU intensive, GSM also is much more packet-loss-tolerant, so if you have network issues or internet issues then GSM will sound a lot better than G729.
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thanks

Postby brett05 » Tue Jan 05, 2010 9:20 pm

yes mflorell you are right
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Postby Michael_N » Tue Jan 05, 2010 9:23 pm

mflorell wrote:In addition to using less CPU intensive, GSM also is much more packet-loss-tolerant, so if you have network issues or internet issues then GSM will sound a lot better than G729.


Are there any licensing fees related to use of gsmcodec?
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Postby mflorell » Tue Jan 05, 2010 9:25 pm

Nope, GSM is pretty much free and clear.
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thanks

Postby brett05 » Tue Jan 05, 2010 9:30 pm

yes GSM is very pretty and he is free and it found as default in asterisk instalation .
we can also make all with GSM if we use a sip for exemple in /etc/asterisk/sip.conf we can just put allow=GSM so here all sip between softphone ans asterisk will be 100% GSM
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