Is Asterisk the solution?

Asterisk Asterisk Support
1

I work at a smallish business. We’ve got an aging Asuzi Pro-616 with four incoming POTS lines. This in turn runs our (also aging) 13 Asuzi multi-line phones. We just advertise the main telephone # and the telco rolls over to the next line(s) automatically when one is busy. Other than the phones that support the multiple lines there aren’t any other special features.

This system, as I mentioned, is aging poorly. The audio on the phones is bad occasionally. Dropped calls (which I’m guessing is the KSU failing). Etc. We’re looking to replace it. The quotes we’re getting seem to be $5k-$6k and up and that’s for a basic system with voicemail. I think we can do it cheaper using Asterisk and get more features than we’ll ever use.

Initially we’re just looking at duplicating what we’ve got with the POTS phones but will possibly switch to VOIP in the future. Basically, what we need is this. When a call comes in on any of the four lines each unused phone rings. Currently a “line #” button lights up so we can place calls on hold and switch lines if we need to. So the new system needs to either replace that or duplicate it. And we’ll be replacing all the current phones obviously. Outgoing calls currently work by just selecting a line and dialing. This is less important I imagine and could be handled automatically. And of course we’d like to take advantage of some of the other nice features of asterisk while we’re at it.

So, obviously I’m new to Asterisk. Is this something it can handle or should we just stick with the proprietary stuff until they decide to upgrade to VOIP in the future? Thanks for any advice or insight you may have. Have a good day!

Alan Barnes
DTS Reprographics, Inc

2

You can definately use * for your business. Get a Digium TDM400 or Sangoma A200 card, and put all the lines in the group. (say group 1)

When you need to dial, just dial Zap/g1/${EXTEN} and asterisk will select the first available line and use it.

in zapata.conf put all the lines in the same context (which goes to your IVR) and all incoming calls will be handled the same. If you don’t want an IVR, you can just do like
exten => s,1,Dial(SIP/1234&SIP/1235&SIP/1236,20)
exten => s,2,VoiceMail(box@context)

which will ring three phones and then go to VM. Personally I’d recommend an IVR as it gives a good business impression, or at least have the line answer with a greeting and then play ringing to caller until answered.

Dialing can be by dialing 9 and getting direct access:

exten => 9,1,Dial(Zap/g1) ; pressing 9 will directly pick up a line, wait for dialtone and dial

or by matching patterns
exten => _NXXXXXX,1,Dial(Zap/g1/${EXTEN}) ; 7 digit local
exten => _1NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN}) ; 11 digit LD
exten => 911,1,Dial(Zap/g1/${EXTEN}) ; dialing 911 is good to have
(with the above just dial a number, no need to dial 9)

Answering calls is much the same, most IP phones can handle 2 or 3 calls at a time.

One thing which asterisk can’t easily do (currently)- if you have a call on line2, and you yell down the hall BOB PICK UP LINE 2, and he pushes L2, that he now is on L2. You have to either transfer the call to Bob, or park the call and have him dial that parking space to grab it. This is called shared line appearance and is scheduled to be included in Asterisk 1.4.

Also for phones- you can get ATAs or FXS ports if you want to save $$, but I’d recommend IP phones. SNOM or AAstra, Grandstream if on a tight budget.

Good luck!

3

Thanks for the quick reply! I’ll definitely have to look more into this. I was looking at the Grandstream GXP-2000’s that you recommended for someone else and they look perfect for us. Thanks again! I’ll probably hit you guys up for some more advice when the time comes.

Alan

4

If you go with the GXP2000, you can probably use the method discussed on the forums for doing shared lines with the GXP.

Part of my office is a small type environment, I have the users answer the phone, press the speeddial button that automatically parks the call, and then anyone can pickup the call by pressing the button that is lit up. It works great on the GXPs. Haven’t tried it on Aastra or SNOM. It takes a few more button presses on the Polycom, and doesn’t work too well at all on the Cisco (Cisco does not support presence, chan_sccp does, but you cannot press a speeddial button that is in use and dial the extension with SCCP).

Take a look at this thread:

forums.digium.com/viewtopic.php? … shared+blf

Wonderful work and works great! We’ll turn asterisk into a key system yet :smile: