Getting a trunk working with CallVoip

Discussion in 'General' started by !Helder, Jun 8, 2010.

  1. !Helder

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    Hello,

    I already set up a telephony server with elastix successfully. For now we used the same provider for all the trunks called Budgetphone.

    Now, i'm trying to make a trunk work with CallVoIP. They have a manual online for configuring Asterisk based systems.

    Callvoip manual. Hope you guys can read this because it's dutch. As you see, you have to change the useragent in sip_general_custom.conf but when i do that, it doesn't really change anything to the system when i check it with the command sip show peer (userid).

    The manual also describes trunk settings which aren't in the right order. Maybe you guy's can give me some tips. I already now that disallow should be above allow.

    current situation... when i'm calling out. i get the famous all circuits are busy message. i isolated the dial rules. I can't test incoming calls because the number is forwarded to another number for the testing environment.

    If you need additional information about the situation just ask me.
    Thanks.

    Donny
     
  2. Patrick_elx

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    from what I read:


    you can try to put that in your trunk peer details:

    disallow=all
    allow=ulaw&alaw&gsm
    authuser= yourlogin
    context =from-trunk
    fromdomain= sip.sipnl.net
    fromuser= yourlogin
    host= sip.sipnl.net
    insecure=very
    nat=yes
    port=5060
    qualify=no
    secret=yourpassword
    type=peer
    username=your login


    and the register string:
    yourlogin@sip.sipnl.net:yourpassword:yourlogin@sip.sipnl.net/yourlogin

    seems strange to me, maybe it should be instead
    yourlogin:yourpassword@sip.sipnl.net/yourlogin
     
  3. !Helder

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    Thanks for your reply.

    I'm going to test this configuration today.
    I let you know if it works.

    Greetz.
     
  4. !Helder

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    It works! :laugh:

    The problem was the register string.
    I'm using username:password@host/username now.

    Thanks for your help
    Greetz.
     
  5. Patrick_elx

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    you can replace the /username at the end of the register string by whatever you want. That's what will be presented when you receive a call, and that what you should put in your inbound route CID for routing.
     

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