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darthanubis
join:2010-01-05
Chesterfield, VA

1 edit

darthanubis

Member

Building a Linux based Call Center for small business IP-PBX

Basically, I tooled around with some voip stuff years ago just trying out stuff in the repositories when I used to distro hop.

But I'm out of my element, and doing massive amounts of reading , and virtual installs.

I would like input from those of you who have setup a "call center" basic IP-PBX system. If you could list the hardware I'd need to grab that'd be wonderful.

Here is where I am so far:

I was going to just start from scratch with a centos or ubuntu server and install apps one by one, asterisk, yate, softphones and such. Then google revealed that there are distros that have already done the heavy lifting for me. Elastix, PBXIAF, Digantel, FreePBXdistro, AsteriskNOW,Trixbox, and Vicidial.

I've installed all of these in a Proxmox VM.

1.Elastix, installed fine, but their help documents were partially in spanish, and I could not seem to get the application to download ANY addons. At least they have an IRC channel, which was not helpful. I worry about support from them.

2.PBX In A Flash, I'm and still trying to successfully install this one. They provide an openvz container, but I can't get it to work. The ISO install did not resolve at the end, it kept trying to restart asterisk. I'm reading good things about this one, so I'd like to at least get it to install. There seems to be a lot of documentation and help on their site.

3.Digantel's dashboard is buggy it seems. The only thing that works is it's FreePBX install. Seems halfbaked.

4.FreePBXdistro ,seems to be just a clean install of Asterisk and FreePBX. It just works. But does not have "call center" features, like for managing agents and such, or I could be wrong, which is why I'm here.

5.AsteriskNOW, Just like FreePBX, just installs and works. You get Asterisk/FreePBX install. No CRM like Sugar or other things I believe I will need, or like to have.

6.Trixbox, I have not installed yet, but from the website, seems to be a competitor to PBXIAF. More than just Asterisk/FreePBX install, CRM etc.

7.Vicidial, This seems to be the whole call center suite as it is dubbed. It installed fine, everything seems to work. The permissions on even the Admin user seems to be restrictive, but I'm sure I need to further read the manual.

I understand there are Digium interface cards, trunks, DIDs, sip accounts, modules... I'm watching some Youtube videos while rebuilding my Proxmox server, and a gateway for the company. Deciding on a UTM/Gateway is more I understand, but I've at least narrowed that down to Zentyal and ClearOS.

So, 4,5, and 7 are winning me over, maybe 6, but the last version of Trixbox released was 2 years ago. I don't know if that a good thing or not?

»community.spiceworks.com ··· how/2394

»forum.voxilla.com/thread ··· p.26454/

The spiceworks link seems like it does what I ask. But your experiences are most welcome.

ropeguru
Premium Member
join:2001-01-25
Mechanicsville, VA

ropeguru

Premium Member

First, you might get better help on this from the VOIP Forum.

»VOIP Tech Chat

I will hey mod this and see if we can get it moved.

Maxo
Your tax dollars at work.
Premium Member
join:2002-11-04
Tallahassee, FL

Maxo to darthanubis

Premium Member

to darthanubis
If I were doing this I would want something with professional support. Trixbox has that. They also have hardware partners that sell turnkey Trixbox solutions.
Perhaps if you called them they could hook you up with someone who can help you get the right hardware for a call center.
darthanubis
join:2010-01-05
Chesterfield, VA

darthanubis to ropeguru

Member

to ropeguru
said by ropeguru:

First, you might get better help on this from the VOIP Forum.

»VOIP Tech Chat

I will hey mod this and see if we can get it moved.

I was concentrating on a FOSS solution. Figured anybody else would just tell me to buy some hardware and plug up. FOSS, and building the server myself is what I'm doing to minimize costs.

ropeguru
Premium Member
join:2001-01-25
Mechanicsville, VA

ropeguru

Premium Member

Certainly not over in the VoIP forum. You will actually get suggestions from both sides. If you want FOSS, they would be happy to help you out.

They just know a lot more over there about what's available and how to set it up as a some of them have done just what you are.
darthanubis
join:2010-01-05
Chesterfield, VA

darthanubis to Maxo

Member

to Maxo
said by Maxo:

If I were doing this I would want something with professional support. Trixbox has that. They also have hardware partners that sell turnkey Trixbox solutions.
Perhaps if you called them they could hook you up with someone who can help you get the right hardware for a call center.

Trixbox looks polished and installed without a hitch, but the software is dated. Their FreePBX version seems stuck at 2.5, and FreePBX is at 2.10. As I tried to update the modules, the UI started to just fall, apart.

Vicidial is what I'd eventually want to deploy. Right now it has a steep learning curve, and it seems that just FreePBX as a frontend to Asterisk, is all anyone needs to get going. All of these distros just seem to be adding around that combination. Two of the simplest of which are AsteriskNOW and FreePBX. Zentyal even has a voip module, which uses again Asterisk.

yaplej
Premium Member
join:2001-02-10
White City, OR

1 recommendation

yaplej to darthanubis

Premium Member

to darthanubis
I just deployed a PBX with AsteriskNOW and FreePBX. Its not been without issues but now that things have been ironed out the owner is pretty happy with how it works.

After they sign-off on the initial install Ill be helping get them setup with a CRM (probably vTiger) and looking for some software call control client. So far Iv found HUDLite but it is dated. Of all the others I found iCE Breaker's Asterisk Call Manager looked the most promising but it seems like its been abandoned without the source ever being published.
darthanubis
join:2010-01-05
Chesterfield, VA

darthanubis

Member

said by yaplej:

I just deployed a PBX with AsteriskNOW and FreePBX. Its not been without issues but now that things have been ironed out the owner is pretty happy with how it works.

After they sign-off on the initial install Ill be helping get them setup with a CRM (probably vTiger) and looking for some software call control client. So far Iv found HUDLite but it is dated. Of all the others I found iCE Breaker's Asterisk Call Manager looked the most promising but it seems like its been abandoned without the source ever being published.

Please keep me posted as to your developments. When you say AsteriskNOW and FreePBX, AsteriskNOW comes with FreePBX as they all do, but FreePBX has there own distro, and they are using iSymphony Call Manager ,Easy to use and affordable call management for FreePBX. iSymphony is a java applet looks great.

I have not looked at ICEBreaker yet.

Have you looked at Vicibox, or GoAutodial?

»www.vicidialnow.org/

»www.vicibox.com/server/i ··· dex.html

FiReSTaRT
Premium Member
join:2010-02-26
Canada

FiReSTaRT

Premium Member

If you install FreePBX as a distro, don't expect any updates any time soon. I have a machine like that, turned off and collecting dust. You're better off just instaling CentOS and * on top of it. That's what I'm planning on doing after I wipe off the dust

XCOM
digitalnUll
Premium Member
join:2002-06-10
Spring, TX

1 recommendation

XCOM to darthanubis

Premium Member

to darthanubis
Your best bet is FreeSwitch for high call volumes even ser or kamilio...
Small call volume go with asterisk.
For the OS. Use Arch.
darthanubis
join:2010-01-05
Chesterfield, VA

1 edit

darthanubis to FiReSTaRT

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to FiReSTaRT
Click for full size
said by FiReSTaRT:

If you install FreePBX as a distro, don't expect any updates any time soon. I have a machine like that, turned off and collecting dust. You're better off just instaling CentOS and * on top of it. That's what I'm planning on doing after I wipe off the dust

Really?

Why did you not just shell in and yum update the machine? I've done that to all of these distros.

I've been reading and now found out about YATE and SIPX. But no one seems to building distros from these. Seems Asterisk is everywhere. Anyone have experience with these servers?

On edit I have found a beautifully clean distro made with YATE

FreeSentral

»www.freesentral.com/inde ··· ome/Home

FiReSTaRT
Premium Member
join:2010-02-26
Canada

FiReSTaRT

Premium Member

I've been doing exactly that for several months and coming up with nothing. That's why I shut her down in the first place.. Too much of a security liability. Just gonna run * on top of CentOS.
darthanubis
join:2010-01-05
Chesterfield, VA

darthanubis

Member

Several months of shelling into a box, running yum, and getting "nothing"? Don't see how that is a security liability. Seems to me, the box is not even connected to the internet, if indeed nothing is occurring. If your getting zero for months,I don't know how you even came up with any kind of conclusion, especially with in regards to security?

FiReSTaRT
Premium Member
join:2010-02-26
Canada

FiReSTaRT

Premium Member

I was able to install packages as needed (which means that it was connected to the Internet as I didn't host a repo on my LAN), but running yum update came up dry each and every time. That's why that particular machine never entered the "production testing" stage.

yaplej
Premium Member
join:2001-02-10
White City, OR

yaplej to darthanubis

Premium Member

to darthanubis
AsteriskNOW also had an option for AsteriskGUI. I saw iSymphony but it cost money and the company I was doing this for wanted to keep everything really cheap. So Im still looking for Call Manager Software. vTiger is said to integrate well from what I have read.
darthanubis
join:2010-01-05
Chesterfield, VA

darthanubis

Member

Have to look at Vicibox/GoAutoDial, either or giving you Vicidial, a complete open source call center? I think that is what I will end up with. I may use FreePBX or Asterisk now at home. YATE 4.0 just dropped so I should try that out.
darthanubis

darthanubis to FiReSTaRT

Member

to FiReSTaRT
Now I understand you.

FiReSTaRT
Premium Member
join:2010-02-26
Canada

1 recommendation

FiReSTaRT

Premium Member

said by darthanubis:

Now I understand you.

I tried out FreePBX because on paper it looked like the best option for a turnkey PBX solution - which was something I wanted to experiment with. Coming up dry with yum updates while being able to pass traffic on the Internet was concerning to me. Maybe I was doing something wrong due to my inexperience with CentOS variants, but there were a couple of features that I was still missing and I was doing more editing of config files than playing with the web interface, so I decided to just run * on top of vanilla CentOS. That's why FreePBX never went from "testing" to "production testing" on my server.

XCOM
digitalnUll
Premium Member
join:2002-06-10
Spring, TX
(Software) pfSense
MikroTik CRS125-24G-1S-RM

XCOM

Premium Member

I really can stress enough how bad is to use this gui asterisk variants.... The first time you run to a real issue you wont be able to do s***t with the "core" asterisk.... and config files are bloated as hell.... Also mind you that you cant move config files around from install to install...
Take the time and learn asterisk. You wont regret it.
darthanubis
join:2010-01-05
Chesterfield, VA

darthanubis

Member

said by XCOM:

I really can stress enough how bad is to use this gui asterisk variants.... The first time you run to a real issue you wont be able to do s***t with the "core" asterisk.... and config files are bloated as hell.... Also mind you that you cant move config files around from install to install...
Take the time and learn asterisk. You wont regret it.

Thanks for the input. I was starting to come to that conclusion myself.
darthanubis

darthanubis

Member

Do I need a fxo/fxs card before I can use my PBX? And must I buy service from a voip provider to make sip calls?

I see places for trunks/DIDs. I just want to test in house before I go the full nine.

All this reading and I still don't understand this part. Maybe because I have yet to setup one of these distros correctly. I keep hearing about a SIP provider. Well, I have an account with ekiga.net, but that is it.

yaplej
Premium Member
join:2001-02-10
White City, OR

1 recommendation

yaplej

Premium Member

If you get a SIP account like from SIPSTATION then you dont need a FXO/FXS card to try Asterisk out. Thats what I did first just got a single SIP line from SIPSTATION to see how things worked like configuring Users and Queues IVRs(Auto-Attendants).
darthanubis
join:2010-01-05
Chesterfield, VA

1 edit

darthanubis

Member

This new foray into VOIP has been an education for sure. At first it seemed everything was based on Asterisk/Digium. Now I have been exposed to Freeswitch, Yate, OpenSER, Sipex, some of which have their own distros. Yate has Freesentral, which from the demo has a clean web interface. FreePBX/Asterisk has a nice layout, but seemingly too many options. Freeswitch has Bluebox which is just a joy to look at and use. Bluebox also has a xmpp module to use Google Voice. You can also switch drivers from Freeswitch to using Asterisk in Bluebox! How cool is that? Yate/Freesentral can use GV as well, but Freesentral's iso had a broken gnome desktop so I could not access the installer. Compiling from source, failed as I could not bring up the web ui as of yet.

»demo.freesentral.com/index.php

Yeah, my apache install does not know what to do with the php extension to the index file. I know, I still have not learned apache.

I digress, Yate, and Freeswitch both seem to be the future and more powerful as far as scalabilty from what I have been reading. With these wonderful UI's, Yate has the excellent Yateclient, jumping onboard with these powerful players is easy,and fun. I can make calls from my PBX through my GV#?!?

My cell phone just so happened to go down, and I'm pinched for cash. So I was especially motivated to get PBX/GV to work. After all the setup and testing in Linux, I boot into Windows only to find installing YateClient, inputting my gmail credentials and one more click gave me the ability to make calls through GV.

As an aside I had to reboot because KDE's graphics are doing some weird shit, kmail was locking because it could not reach the gmail server..etc.

With PBXIAF,it just seemed so hackish, and constant help docs here and there, things working only until a upgrade broke funtionality, and Ward had to write another hack. I can't honestly evaluate PBIAF because I could not get it to run. Maybe on a dedicated box it would run, but I don't have that luxury. if I'm was going to use Asterisk only, I'd use AsteriskNOW or FreePBX. AsteriskNOW is the only distro besides Bluebox I have running with no problems. Since I am just starting to learning VOIP,and IP PBXes, I think it'd be best to go with the one that is the most powerful and built for the fute. I also understand everything has it's place, and Asterisk may just be all a home PBX system will ever need. Yet, knowing that the Big Boys use Sipex/Amazon, and Freeswitch , and
So right now, I'm big on Yate/Freesentral, and Freeswitch/Bluebox. Looking at the traffic on Yate's IRC channel, there does not seem to be in the way of an active community as a resource for help. However, their webpage has a good amount of decently written documentation. Freeswitch seems to be the Asterisk killer, written by a former Asterisk developer. The IRC channel, and website are a buzz.

Softswitches/PBXs, a lot I have learned. Most thankfully to "Eli The Computer Guy's" Youtube series. This guy is great! Even things I feel comfortable with I watched his videos on the topic, and I was enlightened. I watched Eli's class on the Cloud, and it was timely,because now I'm coming across Hosted PBXes like Onsip. Again, being currently conservative in my liquidity, I think I'll opt out of the cloud for now. Cloud=$$$$$$$.

I still think it is cool that I could upload my virtual install of my PBX to the cloud and it'd still work. However, I still need my local one to plug in analog phones, until I can move to all softphones. My top Linux softphone is Twinkle for now, probably soon to be replaced by YateClient.

The professional software that we have access to for FREE, just has me in awe. I installed my first linux distro back with RH5? This brings back the excitement of my Mandrake days! And on edit, Freesentral has a wizard tab, to walk you thorugh the setup.
darthanubis

darthanubis

Member

I forgot Vicidial as a complete Call Center application.

VICIDIAL is a set of programs that are designed to interact with the Asterisk Open-Source PBX Phone system to act as a complete inbound/outbound call center suite.

»www.vicidial.org/vicidial.php
darthanubis

darthanubis

Member

Taking a break from setting up my Sipxec VM, I perused Distrowatch and found an article about Voyager Linux. I check out their website only to see, it is a Debian based minimalist distro, which until recently, produced a VOIP version with Asterisk. I look at the one screenshot, which is of a clean webconfig page, putting FreePBX's to shame. I see over to the left of the UI,a Googletalk button!?! I thought only Yate, and Freeswitch were capable of GV access? Time to fire up another VM. Too bad they stopped making this version of their distro. Although they still have the ISOs up though.
darthanubis

darthanubis

Member

Taking a break from setting up my Sipxec VM, I perused Distrowatch and found an article about Voyager Linux. I check out their website only to see, it is a Debian based minimalist distro, which until recently, produced a VOIP version with Asterisk. I look at the one screenshot, which is of a clean webconfig page, putting FreePBX's to shame. I see over to the left of the UI,a Googletalk button!?! I thought only Yate, and Freeswitch were capable of GV access? Time to fire up another VM. Too bad they stopped making this version of their distro. Although they still have the ISOs up though.

»linux.voyage.hk/files/li ··· risk.png

ropeguru
Premium Member
join:2001-01-25
Mechanicsville, VA

ropeguru to darthanubis

Premium Member

to darthanubis
Google Talk != Google Voice

Google Talk is their chat setup like AOL IM.
darthanubis
join:2010-01-05
Chesterfield, VA

darthanubis

Member

said by ropeguru:

Google Talk != Google Voice

Google Talk is their chat setup like AOL IM.

Cmon guy, I know that. If you setup YateClient to make calls using GV, it does not ask you for your GV#, it asks you for your GT/GM address. It uses the XMPP protocol, which Jabber and GTalk both use to access your GV account and call out. Google has all the crap tied in together.

ropeguru
Premium Member
join:2001-01-25
Mechanicsville, VA

ropeguru

Premium Member

said by darthanubis:

said by ropeguru:

Google Talk != Google Voice

Google Talk is their chat setup like AOL IM.

Cmon guy, I know that. If you setup YateClient to make calls using GV, it does not ask you for your GV#, it asks you for your GT/GM address. It uses the XMPP protocol, which Jabber and GTalk both use to access your GV account and call out. Google has all the crap tied in together.

Apparently not as you said this,
said by darthanubis:

I see over to the left of the UI,a Googletalk button!?! I thought only Yate, and Freeswitch were capable of GV access?

Which is why I posted what I did.

XCOM
digitalnUll
Premium Member
join:2002-06-10
Spring, TX

1 recommendation

XCOM to darthanubis

Premium Member

to darthanubis
To be able to use GV the system has to simulate a client interaction with their jabber network. So in sense it's like using a google chat client. This is the same for Asterisk.