Configuring a SIP trunk


I’m new in VoIP and know very little. I’d like to start playing with Asterisk but have a trouble configuring a SIP tunks now. (I’m using a Trixbox’s Asterisk distribution).

I have a 2 accounts now:

  1. one account on a Mera MVTS server, they only gived me a 10 numbers 1234560 - 1234569 and ip address of the server, also they said that they want to receive and will send all number in full format with country code and asked me for my IP address, so they will authorize me by ip I think. I can oly originate calls from this tribox, when I tr to call to these numbers I hear busy tone and no log lines in Trixbox’s logs appears. Please show me the right way to configure this trunk.

  2. I have also 2 numbers from another provider, there is little bit more of data but I can’t configure it anyway, I have this provider’s instruction for configuring X-Lite softphone, maybe someone kleve could translate it to Asterisk’s rules for me? Here is it:

System Settings -> Network:
Auto Detect IP: Yes
Out Bound SIP Proxy:

  1. Press BACK -> SIP Proxy -> [Default]:
    Enabled: Yes
    Display Name: Your name
    Username: number
    Authorization User: number
    Password: ********
    SIP Proxy:
    Out Bound Proxy:
    Use Outbound Proxy: Always

  2. Press BACK until menu’s top and ?

Advanced System Settings -> Audio Settings -> Silence Settings:

Transmit Silence: Yes