Starting Configuration of asterisk

Hi all,

I have just started working on asterisk. I installed asterisk easily on red hat through
make install

Now, for configuration of asterisk, i need to add accounts of users in sip.conf and need to make dialplan in extensions.conf

I want to ask where these sip.conf and extensions.conf are kept. It’s said they are kept in /etc/asterisk. But, after installation, there’s no such file in /etc/asterisk.

So, what should i do?
Do i need to make sip.conf, extensions.conf by own my own and keep them in /etc/asterisk ?
What about asterisk.conf ?
where this file resides, is it necessary to keep it in /etc/asterisk ?

One more thing, i’d like to ask, what IP addr asrterisk uses for itself. Does asterisk uses IP addr of machine that is 192.168.x.x tha can be seen thru ifconfig.

Please help,
I need help urgently.

Thank you all.

at the end of the installation of asterisk it asks you if you wan’t samples you did not read it?
go back to the directory with the asterisk source and type
make samples
this will create all files in /etc/asterisk

Hi all,

When i run :
make samples

It has shown all .conf files in /etc/asterisk where i have got sip.conf and extensions.conf

and i have made my accounts entry in sip.conf and basic dial plan in extensions.conf.
one small query :
Do i need to add general info(like type,username,secret,host, etc …) of users or it’s necessary to add this line too :
register => 123:pwd@192.168.x.x(asterisk ip):123

But, now my big question arose when my X-lite client makes call to another client, it keeps on connecting in xlite window and then timeouts,
and there is no reponse on asterisk server. What does that mean xlite is not connected to asterisk. But in xlite user settings and network, i have provided asterisk machine IP address 192.168.x.x in xlite’s filelds - SIP proxy, outbound proxy and domain/realm.

I think it should connect, but it’s failing. I might be providing wrong asterisk address.
I want to ask whether asterisk gets any different IP address when it runs and that IP address is different from ip addr of machine on which asterisk runs.

Please help.
Waiting for response.

Thanking you all.


you need 3 elements to make a sip line show up and be callable in *.

  1. users.conf is used to define the address and attributes of each extension
  2. sip.conf is used to establish how the extension registers and what context it is in by default.
  3. extensions.conf is used to establish the DTMF sequence(s) that the phones use to speak to each other.

The CLI interface can show you when things are correct.
sip show peers will tell you that the line is registered

dialplan show default will tell you if you can dial it.