Some newbie advice plz

Asterisk Asterisk General
1

Hello dear forum members,

i’m rather new to asterisk and telephony in general. I have to do quite a large project, which will be involving asterisk. Maybe a small introduction would be nice…

I work for a company with several call-centers. At the moment we use “commercial” solutions for our outbound calling. Expensive pbx soft\hardware and a really expensive predictive dialer (1500 euro’s a seat!). Cause we’ve got about 200 seats, we’d rather not pay 1500*200 (as u might imagine). Anyway, we’ve been looking around to replace our stuff and found asterisk, which we think is AWESOME! Anyway, cause i’m rather new to all this stuff (don’t work here that long) and i’m going to roll out this project, i really am in need for some information! So, if anybody has some answers to my questions that would be great!

Ok, the following is what i have in mind for this project (questions in there as well) : And excuse my english here and there, not a native speaker…

We’ve got several call centers, as mentioned. We’d like to connect all of them to one fat pbx system, or several interconnected pbx systems. At the moment we have analog old skool phones. Would it be wise to continue using those, or is it better to use voip internal with regular phone lines to the outside? What is the advantage of using voip inside against analog (except for the wiring)? And, if we’d use analog lines, do i need a digital E1 card for the outside world and an analog card in my pbx? Or can i also use voip with my analog phones, by hooking those up to a analog card that u can buy at digium? Some hardware advice would be highly appreciated. Cause, as mentioned before, i’m not a phone hardware master (yet;)).

Is it possible to connect pbx-s from several buildings, so we have one “big” pbx? Or can i just build an ultra fat pbx and connect all the buildings trought some vpn with it? I noticed u need quite some big hardware to get good quality on concurrent phone calls. So, maybe clustering it would be a better idea, since we might have 150 or more concurrent calls at a time?

Oh yeah, not really un-important: what distro is advised? I assume it’s best to run the box without the qui right? Saves recourses…

Then the part of predictive dialing. I’ve seen gnudialer, but that is not totally covering our needs. It’s planned to build that one ourselves. Cause this company uses mainly m$ products, that stuff will be coded in c#. Is talking trough the sockets the way to go? Cause i can imagine with dialing 100s of phone numbers a minute, asterisk might not find that amusing. Does anybody have any experience on that?

Geez, i realize it’s quite a long post… Well, sorry… Just so full of questions and really enthusiastic to get going with asterisk. Main question of this post is actually: analog or voip (internally)? Cause i first have to decide what hardware to buy, then build a test environment, code some api stuff and implement it. Oh yeah, if i asked anything dumb in this post, don’t hesitate to call me a dumb ass… i can live with that, as long u give me some urls to read;)

tnx!

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  1. Good luck on the project.
  2. I would switch over to IP phones. Even with the most basic of phones the functionality that you will get from the IP phone will be worth the investment. Also you will need an FXS gateway which will run you about $40.00-$50.00 (I know in the US) per port. For just a bit more you can get an IP phone.
  3. If you want to connect an E1 directly in to your asterisk server you will need a card for it. The card would be used to connect you to the phone line and has nothing to do with your IP phones. Asterisk lets you “mix and match”. In fact you can set asterisk to use VOIP to call say the US and your E1 to call the UK.
  4. I have personally had a lot better experience with Sangoma cards over Digium cards. While I have not purchased a Digium card in a while from speaking to friends recently there are still some issues.
  5. VOIP tends to be a lot cheaper but some times you run in to a call quality issue. With an E1 you are sort of guaranteed a better connection. Over all I think VOIP has come a long way and if set up correctly it can work pretty well.
  6. I think that it would be more advisable to have PBX at each location. This way if there is an internet issue you can rely on a TDM connection at each location. You can connect all the systems with VOIP or a point to point PRI.
  7. In regards to a distro it’s a matter of preference. I personally like CentOS but again you have to use what works for you. Best to run it with out a GUI (the OS) it just takes up resources and how often will you really use it ?
  8. There is very good software out there for Asterisk (as far as a call center). I can’t recall the name now (PM me and when it comes to mind I will post it here) but I have been told that it works great. It has an autodialer built in. The amount of calls that asterisk can send out at a time depends on the type of box you have, if you are doing any trans coding, recording etc.
  9. Start off with a test box and use VOIP. It’s a lot cheaper. There are several free soft phones out there. You can also get a cheap VOIP line to test incoming and out going out of your box.
  10. Again good luck.
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Thank u very much Dovid, for your very useful answer!

So, if i’m correct, i can also make a virtual testbox, without the hardware? Meaning that if i install asterisk in a virual box on my own machine, i can hook it up with a softphone and an outgoing voip line? That would be awesome and great for testing purposes, cause i don’t have to get me extra hardware in my already cluttered office. And if that is the case, it will make testing interoperability easier as well. Just have to get me some extra ram then;)

I’ve checked out the sangoma website and have to say it looks promising. Tnx for that advice!

I’ll dive into it today! Excited to get started!

and tnx again for the time you spent in answering my questions (and not calling me a dumb ass;))

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  1. I rarely rip on Noob’s since I was also a noob at one point.
  2. Running Asterisk on a virtual machine may give poor performance. For testing 2-3 calls get an old p3 500 mhz and load up asterisk on to it. You don’t need any hardware to test. I know VOIP companies that all they have is servers. No cards what so ever.
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tnx for sparing the n00b;)

i’ve managed to set up a voip server on a vm and can call from machine a to machine b. sweet!

i have a server, so gonna set that on up soon to do some “real” testing. For the meanwhile i just test making connections from a to b. At least i can make a start coding and actually see what my code does, instead of just guessing.

If a my ask just one more question: i noticed that if i add another user, i have to restart asterisk to be able to log in with the new user. Is that normal?

I’m browsing sourceforge at the moment and see there’s loads and loads of apps for asterisk. Awesome!

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  1. If you add another user to sip.conf you can just reload the sip files by typing in the cli “sip reload” (with out the quotes). To reload all the configuration files simply type in reload.
  2. I would stay away from GUI’s if you can. Messing around with just “regular asterisk” is the best way to learn IMHO.
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ok, so just reloading the files does the job? great! i read about using a database backend for asterisk. Does that mean all the .conf-ing is loaded into the database? And if so, does that have any advantage over files?

2: i didn’t find anything useful for this particulair project on sourceforge anyway, expect for asterisk.net, which i’d like to use, so i don’t have to rewrite everything… As in many projects this one should be finished yesterday, so using an existing class really helps speeding up the process i guess. and i’ll try doing as much as i can trough telnet