Kevin McAllister on 5 Jun 2011 10:32:18 -0700

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Re: [PLUG] VoIP setup

Hi Julien

On Jun 3, 2011, at 3:58 PM, Julien Vehent wrote:

> Hey guys,
> I'm looking for advices regarding VoIP products. I need to build an in-house VoIP PBX for one of my company's office, and while I'm fairly sure we will go with Asterisk (still leaving the door open to OpenSIPS), I'm not certain I want to build it myself, or acquire an appliance.

If you're just processing SIP traffic, OpenSIPS or Kamillio are the way to go, but if you want features (voicemail, conference bridge, call queueing or ACD functionallity) you're going to want something like Asterisk or FreeSwitch (I think).  I know you can probably pull some of that off with OpenSIPs but it will be harder.   

> We want around 30 parallel lines, and do the convertion VoIP to analog __before__ it leaves the office (no hosted PBX or anything like that, we don't trust the network bandwidth enough, it's just a basic DSL there).

So I read other posts in this thread and I see this is a call center.  The real question I'd have is outgoing or incoming calls?  And how busy?

Going all SIP makes a lot of sense from an upfront cost and flexibility perspective but there are other factors to consider.  If you're taking mostly inbound calls on POTS lines or a PRI and it's not mostly to a toll free number, it'd probably be cheaper to not go SIP.  In most cases you're going to pay per minute on SIP calls, but in traditional telephony they charge you for the line and all the incoming minutes are "free."

I think I saw later in the thread something about 10¢/minute.  No matter what you do, you should be able to do quite a bit better than that.

> I've looked at Digium, they have interesting prices. But does it leave the door open to customization ? Do you have access to the database, the logs, the system in general ?

If you are talking about the SwitchVox product line, it's pretty nice but there are a few things I found confusing when I first heard of them:
1. seat licenses.  There was some cost per phone they would provision you buy tokens or something (this may have changed in the two years since I last looked at them)
2. Getting under the hood was not supported.  The device was all sorts of locked down and you weren't supposed to go in there to do anything, everything was to be done through their web interface.  

That said my employer has a few customers that use them as SIP trunk customers and they really like the easy of changing call flow and stuff.  I assume switchvox probably have export of call records too so you can do your own sorts of reports.  They also support some sort of API so when a call comes in it'll hit a URL for something and you can feed it back things to do based on some other software you have.  I thought it was all pretty slick, but it isn't something I've worked with much at all.

Regarding hardware, if you build your own asterisk box you can either buy cards from digium or sangoma (I came to like the sangoma cards better for connecting T1-PRI when CoreDial still did that sort of thing in the datacenter) That can allow you to plug in POTS lines or a T1-PRI (equivalent to 23 POTS channels each)

I assume there is some sort of little device that has a bunch of FXO ports you can use to convert POTS to SIP that may end up being cheaper than cards.  Maybe the Adtran can even do that the 908e that sits in my office has an Amphenol connector ( on it.

Ideally if you want to try out Asterisk I'd recommend doing a simple prototype because there are going to be factors you won't even think of until you put something together and have someone actually use it. (Is incoming caller ID important? What about blocking outgoing caller ID? telephony has a ton of little features) 

I'd say download one of those easy to setup Live Disks with a web interface on top of asterisk (FreePBX and possibly Digium has one) use a softphone and pickup some sort of ATA that has an FXO port you can plug a line into, or get a cheap digium card.  Check they're in Manayunk, have always responded favorably when I had a crisis and needed a new card immediately, we've always gotten pretty good prices on them on new stuff.  And who knows maybe they have working/questionable returns or whatever that they can't even get rid of that sit in a box in the office that they'd give you cheap.

> I know the folks at coredial have hosted solutions, anything deployable in house ?

CoreDial doesn't have anything specific to deploy in house, they do support sip trunks so you can bring your own asterisk box and sign up for sip trunking.

I wrote a bunch of stuff, not sure if I answered any questions.  Best.

- Kevin
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