Newbie needs help

My boss told me to look at this Asterisk system and it looks great. But after a whole day of researching i still can’t fully grasp the hardware configuration of the system.

We are a small company of 6, we currently have a old PBX system, Intranet, a server. Can anyone suggest the hardwares that I need to get? As a start I don’t think my boss would want all that much function. Just a system that will replace the PBX, and maybe VOIP.

If possible a architecture diagram would be great.


ok after some more research I’ve came up with a configuration which I have no idea if i am correct or not.

Analog phones are connect to a network switch via an ATA adapter. The network switches can also have soft phones, and SIP phones. The network switch is then connect to the Linux Asterisk PBX which is then connect to the PSTN.

Is this right by any chance?? and What i need to know is what interface card do i need for the server? something like the Digium TE120P?

Please advice as i am lost in blue…

With 6 phones and a call ones in a while you don’t need that much hardware. I have an Asterisk server running on a Nokia 770 and I think it will do in your situation. A normal PC or a not that heavy server will do.

You can use sip phones like Linksys Snom etc. or you can use the anolog phones available with using an ATA (like sipura)

With 6 phones you just need a very basic dial plan. Just make sure lines can come in and you can dial out. Below is an example of an Asterisk for beginners demo.

  • a couple of phones
  • a context for each phone
  • every phone has its own context included with priviliges

If you understand the dial plan below you can use it to brew your own dial plan for your businesss. Your boss will be happy. If you want the real stuff with operating panel, webenabled configuration and monitoring of every relevant aspect spend some money on a ScopServ licence, the best Asterisk gui available. If you have some questions post them and I will answer your questions.



[5000] ; receptie
; siemens
include => plaza_alle_rechten ; plaza all rights

[5001] ; ; afdeling registratie registration department
; e61
include => plaza_intern
include => plaza_outbound_nl
include => plaza_geen_autorisatie plaza no autoristion

[5002] ; afdeling voorlichting communication appertament
; zyxel met riempje
include => plaza_alle_rechten plaza all rights

[5003] ; afdeling financi√n financial department
; zyxel zonder riempje
include => plaza_intern
include => plaza_geen_autorisatie no authorisaition

include => plaza_alle_rechten ; IAX2 idefisk user vmware op macbook

include => plaza_intern
include => plaza_outbound_nl
include => plaza_outbound_int
include => plaza_geen_autorisatie

;5001 = afdeling registratie registraton department
;5002 = afdeling voorlichting communciation department
;5003 = afdeling financien financial department
exten => _5XXX,1,Dial(SIP/${EXTEN},20,rt)
exten => _5XXX,n,GotoIf($[ “${DIALSTATUS}” : “CHANUNAVAIL”]?iax:hangup)
; test for dialstatus. Bij “CHANUNAVAIL”
; naar IAX2 dial regel springen

exten => _5XXX,n(iax),Dial(IAX2/${EXTEN},20,rt)
; call via IAX2 bacuase sip number was not available

exten => _5XXX,n(noanswer),n,Playback(plaza/niet_beantwoord)
exten => _5XXX,n,Goto(hangup)

exten => _5XXX,n(hangup),Playback(plaza/geen_autorisatie)

exten => _5XXX,n,Hangup() ; ophangen, geen extensie (toestel) beschikbaar

exten => 6000/5000,1,Answer()
exten => 6000/5002,1,Answer()
exten => 6000,n,Goto(recording,s,1)

#include “extensions.d/*.conf”

exten => 030xxxxxxx/_X.,1,Answer()
exten => 030xxxxxxx/_X.,n,Playback(plaza/welkom)
exten => 030xxxxxxx/_X.,n,Goto(inbound_menu,s,1)

exten => 0307114197/_X.,1,PlayBack(beep)
exten => 0307114197/_X.,n,Wait(2)
exten => 0307114197/_X.,n,Goto(1)

exten => 0307114198/_X.,1,Authenticate(0621830837)
exten => 0307114198/_X.,n,Goto(recording,s,1)

exten => s,1,BackGround(plaza/menu) ; kies 1 voor de receptie, 2 voor afdeling registratie, 3 voor voor afdeling voorlichting
; en 4 voor afdeling financien

exten => 1,1,Dial(SIP/5000,10,rt)
exten => 1,n,Playback(plaza/niet_opgenomen) ; didn’t pik up call
exten => 1,n,Hangup()
;afdeling registratie
exten => 2,1,Dial(SIP/5001&IAX2/5004,10,rt) ; telefoon rings 10 seconds
exten => 2,n,Dial(SIP/5001&SIP/5002&SIP/5002,10,rt)
; all phones ring 10 seconds
exten => 2,n,Playback(plaza/extern_doorverbonden) ; external transfer
exten => 2,n,Dial(IAX2/Voopxxxxx@voop_plaza/06218xxxxxx,10,rt)
; extern doorschakelen, pas op mailbox mobiel
exten => 2,n,VoiceMail(5001) ; record message
exten => 2,n,Hangup()
;afdeling voorlichting
exten => 3,1,Dial(SIP/5002,10,rt) ; phone rings 10 seconds
exten => 3,2,Dial(SIP/5001&SIP/5002&SIP/5002&SIP/5004,10,rt)
; all phones rings 10 seconds
exten => 3,3,Dial(IAX2/Voop50608@voop_plaza/0621830837,10,rt)
; external transfer, pas op mailbox mobiel
exten => 3,4,VoiceMail(5002) ; record message
exten => 3,5,Hangup()

; afdeling financien
exten => ,1,Voicemail(5002)
exten => 3,2,Hangup()

exten => i,1,Playback(plaza/ongeldige_invoer); invalid input
exten => i,2,Goto,(s,1) ; back to menu
exten => t,1,Playback(plaza/totziens) ; time out goodbuy
exten => t,n,Hangup

exten => 9,1,VoiceMailMain() ; access to voicemail application


exten => _00ZXXXXXXXX,1,Dial(IAX2/Voop50xxxx@voop_plaza/31${EXTEN:1},20,rt)
exten => _00ZXXXXXXXX,2,Hangup()

exten => _000Z.,1,Dial(IAX2/Voopxxxxx@voop_plaza/31${EXTEN:1},20,rt)
exten => _000Z.,2,Hangup()

exten => _X.,1,Playback(plaza/geen_autorisatie)
exten => _X.,n,Hangup

thanks for the reply…

so u r saying that my hardware setup is too complicated? u said a not so heavy server, i have that. What else to do need? i should be going with the Analog phones so i will need the ATAs rite? after the phone line has been plug in to the ATA, where does it hook up to?? do i need a interface card?? and what hooks up to my phone jack on the wall??? -_- going insane…

thank you

If you are set on using analog phones, you have a few choices:

  1. You can get an ATA for each analog phone. The analog phone connects to the ATA and the ATA connects to your intranet. That way the analog phones will appear as SIP phones to the Asterisk server which also should be connected to your intranet.

  2. Instead of ATAs, get a Digium AEX800 standard PCI card.
    Both cards take two S400M FXS Modules which will enable you to connect up to 8 analog phones to the card. Since the PCI card is installed on the Asterisk server, you connect your analog phones directly to your Asterisk server without the need for ATAs.

Hope this helps,

thanks for the great help! i think now i have a board sense of whats going on now…

however there is still one question that i am not quite sure of… does the PSTN goes to the Asterisk server’s modem?

The TDM800 (and TDM400, TDM2400) etc. analog line cards can have a mixture of FXO modules which support the incoming exchange trunk lines, and FXS modules which connect to normal analog phones.

You can also get ATAs that support the trunk lines, but in my own experience these don’t work nearly as well as a TDM card, though the type for extension phones works fine.

The whole system can work with nothing but conventional phones, but you will be losing out on many of the facilities that Asterisk can give.

The digital phones available provide many more facilities, and although somewhat more expensive you can cancel part of the cost against the ATA or FXS module as they just connect to the Asterisk system via ethernet.

One of the most obvious features is having indicators/buttons that show who’s on the phone and which lines are in use.

I use Polycom IP501 or IP601 phones. There are a number of makes, it’s very much down to price vs. quality and features.

Have a look at what’s available before you make any decisions; I just did a quick search and found a place with a good selection:
(No connection, just first site that came up).

thanks for the great help! much appreciated! this indeed is a great community.