Basic configuration

Posted on 2006-04-29
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.


Question by:MarianoSB
    LVL 36

    Expert Comment

    This page gives the basic configuration required :-

    Basically add the following to your sip.conf :-


    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.

    Author Comment

    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 '' - Username/auth name mismatch", asterisk is running in a virtual machin, with ip, i have an adsl router configured with an ip ( the port 5060 is configured right too, with the, my machin has an ip, 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)
    password:1234 < - ¿is this the pass to connect to voipbuster or to asterisk?

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

    //**----------------------------------------------------------------------- my sip.conf-----------------------------------------**//

    bindport=5060                  ; UDP Port to bind to (SIP standard port is 5060)
    bindaddr=            ; IP address to bind to ( binds to all)
    callerid = Unknown
    register => <voipbusterusername>:<voipbusterpass>

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


    //**********-----------------------------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 ?


    LVL 36

    Accepted Solution

    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 :-


    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.

    Author Comment



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

    externip = ;(substitute your public ip address)
    localnet = ;(substitute your lan subnet address)

    in the sip_nat.config


    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.


    LVL 36

    Expert Comment

    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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