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Posted on 2006-04-29
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Last Modified: 2012-06-27
I need help with my asterisk configuration.

I cant find a decent configuration, to use my voipbuster service ( i can use they software very well)

i need a basic sip.conf, or iax.conf and extension.conf

im behind an adsl router.

PLEASE HELP!!


Thanks
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Question by:MarianoSB
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Expert Comment

by:grblades
ID: 16571861
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.
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Author Comment

by:MarianoSB
ID: 16572785
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.

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Accepted Solution

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grblades earned 1200 total points
ID: 16572872
It looks like it is not your connection to voipbuster but the connection between the phone and asterisk itself which is the problem.

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

These entries include the contents of these other files within the current configuration file. You need to check what is in these files. I would expect to see an entry for the SIP phone configuration looking something like :-

[mariano]
type=friend
secret=1234
username=mariano
host=dynamic
context=voipuk
contect=sip

The username/secret here must match what you configure in the software client.

The software client then connects to asterisk using the authorisation in this section and sends the number it wishes to call. Asterisk then looks through extensions.conf to see what should be done and should if configured correctly send the call out via voipbuster.
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Author Comment

by:MarianoSB
ID: 16572964

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
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LVL 36

Expert Comment

by:grblades
ID: 16572989
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.
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