Basic configuration

This page gives the basic configuration required :-
http://www.voip-info.org/wiki/view/Asterisk+VoIPBuster

Basically add the following to your sip.conf :-

[voipbuster]
type=peer
host=sip1.voipbuster.com
username=YOURUSERNAME
fromuser=YOURUSERNAME
secret=YOURPASSWORD

And the following to your extensions.conf :-
exten => _00.,1,Dial(SIP/${EXTEN}@voipbuster)

When you dial a number starting with 00 it will be sent across voipbuster.

Hi, thanks for your post, i try that but my problem seems to be my conection with asterisk.
I've try it with x-lite, and Express Talk, and nothing.

When i look in the asterisk log, i get "chan_sip.c: Registration from '' failed for '192.168.1.6' - Username/auth name mismatch", asterisk is running in a virtual machin, with ip 192.168.1.2, i have an adsl router configured with an ip 192.168.1.1 ( the port 5060 is configured right too, with the 192.168.1.2), my machin has an ip 192.168.1.6, and ther is  conection between all,  it looks like a configuration mistake in asterisk, but i cant solve it.

my Express Talk softphone is configured this way:

full "Friendly" Display Name: mariano
Sip account numer ( or user): mariano
Server(Sip server or Virtual PBX) 192.168.1.2
password:1234 < - ¿is this the pass to connect to voipbuster or to asterisk?

i also include
register => <voipbusteruser>:<voipbusterpass>@sip1.voipbuster.com
in the [general] of sip.conf, like is proposed in many places.

//**----------------------------------------------------------------------- my sip.conf-----------------------------------------**//
[general]

bindport=5060                  ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0            ; IP address to bind to (0.0.0.0 binds to all)
disallow=all
allow=ulaw
allow=alaw
callerid = Unknown
register => <voipbusterusername>:<voipbusterpass>p@sip1.voipbuster.com

#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf
#include additional_a2billing_sip.conf

[voipbuster]
type=peer
host=sip1.voipbuster.com
username=<voipbusterusername>
fromuser=<voipbusterusername>
secret=<voipbusterpass>

//**********-----------------------------end of sip.conf---------------------------------**********//

in my extensions.conf i added

[buster] // is this the context of the user???
exten => _00.,1,Dial(SIP/${EXTEN}@voipbuster)

//**********-----------------------------end of extensions.conf---------------------------------**********//

for the reccord, im usign asterisk@home, and something else that i cant undesrstand is why when i use freepbx, to add sip trunks, etc, i cant see any changes in the sip.conf

where is the place in asterisk to add new user / password ?

Regards.

HEY

That was the solution, i added that to the sip.config, AND added

externip = xxx.xxx.xxx.xxx ;(substitute your public ip address)
localnet = 192.168.1.0/255.255.255.0 ;(substitute your lan subnet address)
nat=yes

in the sip_nat.config

WORKS PERFECT

but the KEY was... RESTART THE SERVER!!! just that..

Thanks for your post. the point are yours

and the softphone is X-LITE, its easy to use. In a few days i will try to configure starshop, a call shop billing software to asterisk, i hope i can count with your support.

Regards

Mariano

Personally I prefer to use the DIAX IAX based client. It has a much more intuitive interface and being IAX based it has less issues with firewalls so is far more likly to work from hotel rooms etc...

I haven't used any billing software. I just wanted something to give basic call stats and costs for outgoing calls from my company. Most software seemed over complicated for my needs so I ended up just writing a simple program myself.