group=0
context=from-pstn
channel => 11
callerid=
group=
context=default
;;; line="12 WCTDM/2/3"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 12
callerid=
group=
context=default
The core of the asterisk configuration is dialplan. Dialplan manage how asterisk handle all the
incoming and outgoing call. It can consist of 3 lines but also can reach tenth or hundreds lines, depends
on how the complexity of our configuration. We can also use macro feature on asterisk. Dialplan is
placed on /etc/asterisk/extensions.conf. My extensions.conf manage how the incoming call should be
handled, how to make outgoing call to PSTN, GSM line and sip extensions, how to make conference
call, how to connect to other asterisk server using IAX2 protocol, use the monitor application to record
the conversation and how to make greeting. I will explain our extensions.conf in more detail in the next
post together with sip.conf, iax.conf, meetme.conf and voicemail.conf.
PART 3
To enable asterisk to communicate with PSTN lines we should have either a VOIP-PSTN gateway or
FXO card. I will not explain about VOIP-PSTN gateway, there are some service providers out there
who provides this service for their customers. In my work I use Digium TDM 410P with 4 FXO port
per card. There are some alternatives in the market like Sangoma, Rhino, etc, the important is we
should make sure that it works with Asterisk either with dahdi driver or zaptel/zapata driver. Also if
possible select the card that already has hardware echo-canceler. Echo is a problem in voip
communication, and if you have card with no echo-canceler than your server CPU will busy do the job.
Just remember that Digium cards are no longer use zapata driver, and some changes has been made to
the configuration file name and location, /etc/zaptel.conf become /etc/dahdi/system.conf and
/etc/asterisk/zapata.conf become /etc/asterisk/chan_dahdi.conf
In the client site you can use any SIP client hardwares or softwares. Ekiga and Emphaty are the good
choice for you who prefer GTK libraries and KCall and KPhone are for you who prefer Qt libraries. IP
phone hardware now widely available in the market from cheap to high price, you can select any brand
as long as it compatibles with Asterisk. In this project I choose Polycom IP-330, I also used
Grandstream and Aastra in other implementation. In this implementation the owner also ask me to use
Polycom KIRK Wireless Server 600V3 with Polycom DECT Handset 3040.
Now the time for the dialplan, extensions.conf, which is the core of asterisk implementation, as an
example let me introduce you with my configuration. It is a good habit to always backup default
asterisk configuration, and start the new configuration from the scratch.
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