VoIPowering Your Office with Asterisk: SOHO VoIP, Part 2

In part one, we learned how to connect an Asterisk server to legacy phones and phone service. Today we’re going to set up a connection to the outside world and set up internal extensions, so we can actually place and receive calls through Asterisk.

Starting the TDM04B at Boot
We need to finish configuring the TDM04B. Last time, we left off with manually loading the drivers for the TDM04B, just to make sure they would. If there are errors in /etc/zaptel.conf the drivers won’t load. Getting the drivers to load at boot is easy. Go to your Zaptel source tree (see VoIPowering Your Office With Asterisk: Moving to the Grownup Version) and install the startup files:

# cd /usr/src/zaptel-1.2.6

# make config

# lsmod

Module         Size Used by

ztdummy         3924 0

wcusb         20000 0

wctdm         34880 0

wcfxo         13088 0

^20zaptel21^        27936 0

wct1xxp        19488 0

^24zaptel25^        65600 0

tor2          91936 0

zaptel        206852 12

ztdummy,wcusb,wctdm,wcfxo,^31zaptel32^,wct1xxp,^35zaptel36^,tor2

crc_ccitt        2113 1 zaptel

Configuring the Server
OK, then, enough messing around with the TDM04B. Time to set up Asterisk to answer and route incoming calls.

The next file to edit is /etc/asterisk/zapata.conf. Start over with an empty file:

# mv zapata.conf zapata.conf.old

# nano zapata.conf

[channels]

context=pstn-test

signaling=fxs_ks

language=en

usecallerid=yes

echocancel=yes

transfer=yes

immediate=no

group=1

channel => 1

channel => 2

channel => 3

channel => 4

Then create the pstn-test context in /etc/asterisk/extensions.conf. Add this to the sample file:

[pstn-test]

exten => s,1,Dial(SIP/250,30)

exten => s,n,Voice-mail(250)

exten => s,n,Hangup

Make sure that you have “TRUNK=Zap/g1” in extensions.conf.

Adding Users
You probably want incoming calls to actually ring a phone, and possibly even be picked up by a human. There are two steps to setting up a new internal extension; both sip.conf and extensions.conf must be edited.

First, register the user in sip.conf, with the user’s real name and IP. Start over with an empty file, then add these lines:

[general]

port=5060

bindaddr=0.0.0.0

context=default

disallow=all

allow=gsm

allow=ulaw

allow=alaw

[alrac]

type=friend

username=alrac

secret=[alrac’s password]

host=[phone IP]

context=pstn-test

The next step is to register the users’ extensions in extensions.conf. Add these to the pstn-test context:

[pstn-test]

exten => 250,1,Dial(SIP/alrac)

exten => 251,1,Dial(SIP/stan)

exten => 252,1,Dial(SIP/ollie)

;Internal users can call each other directly with their 3-digit extensions:

exten => _2XX,1,Dial(SIP/${EXTEN},30)

exten => _2XX,n,Voice-mail(${EXTEN})

exten => _2XX,n,Hangup

Installing Phones
Every IP phone is configured little differently. These are the main parameters you’ll need, as this example for Alrac shows:

Enable: yes

Username: 250

Authorization User: 250

Password: [alrac’s password]

DomainRealm: [Asterisk server IP]

SIP Proxy: [Asterisk server IP]

The Story So Far
We’re not done yet, but this is a good place to stop and review what we’ve done so far:

• Installed and configured the TDM04B card for interfacing between the PSTN and our IP phones
  
• Created a new context, pstn-test
  
• Incoming calls will all be routed to alrac
  
• Calls will automatically go to the first available line
  
• Registered our users in sip.conf
  
• Registered telephone extensions in extensions.conf
  
• Setup IP phones and registered them with the server

Outbound Dialing
Receiving calls is nice, but you probably want to call out as well. We’ll get to this in part three, as there are some perils and pitfalls to be avoided. The biggest one: you must define your contexts and call routing carefully or you risk leaving yourself open to providing free long distance to the world.

We’ll also define a better default context and, in part four, we’ll set up a digital assistant so that Alrac doesn’t have to answer everyone’s calls. Digital assistants are very smart and will even screen your calls.

Resources
Asterisk: the Open Source PBX
Voip-Info.org

Adapted from voipplanet.com.