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Thread: Trixbox Asterisk Help

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  1. #1
    Join Date
    Sep 2008
    Location
    Southwest MO
    Posts
    219

    Default Re: Trixbox Asterisk Help

    Quote Originally Posted by bigdessert View Post
    I'll be truthful, this is all over my head, I will start reading and playing around and see if I can make things work. The one statement that stands out to me here is " I have not played with TrixBox since they went a bit over the edge a while ago." can you please explain what you mean by this? I thought trixbox was the defacto standard open source setup for getting asterisk up and running. Maybe I am wrong. I am just starting to dabble in these setups because the company I work for wants me to start looking into voip phone systems instead of straight digital "nortel" type systems. Yes I am fresh fish and am just starting to get wet, but dont want to head down the wrong direction....I think what you are referring to is the buyout by fonality if I'm not mistaken.

    Thanks for all the help.
    I started out on Trixbox several years ago and it was great. Over the years in my opinion they have become kind of a mess and have had some controversy. I switched over to PBX In a Flash and it has been great. I would recommend taking a look at that. It is similar to Trixbox as in it includes everything you need to get set up, but I think it is easier and less bloated. Some will also say your privacy is more protected..

    Also, as someone already mentioned, you might be better off testing and learning with a VOIP provider that supports BYOD more openly than VOIPo does. I love VOIPo and their service and support is fantastic. I'm a big fan, but wouldn't be my first choice for learning Asterisk. There are a few pay as you go providers that support Asterisk and have good configuration settings and things like that. I don't know if it is appropriate to mention here but PM me if you like. The VOIP Tech Chat forum at broadbandreports.com is also a great resource.
    Scott

  2. #2
    Join Date
    Dec 2008
    Posts
    13

    Default Re: Trixbox Asterisk Help

    I had no problems at all gettting * talking to the BYOD server -- the only time I plugged in the shipped adapter was when I initially got it to be able to get my SIP credentials, and then for some testing for a couple days.

    Here's the FreePBX trunk config I used:

    Dial rules:
    Code:
    911
    1NXXNXXXXXX
    1+NXXNXXXXXX
    1763+NXXXXXX
    Outgoing:
    Code:
    username=763NPXXXXX
    type=peer
    session-timers=accept
    session-refresher=uac
    secret=as3cr37
    rfc2833compensate=yes
    qualify=5000
    nat=no
    insecure=port,invite
    host=sip.voipwelcome.com
    dtmfmode=auto
    disallow=all
    context=from-trunk
    canreinvite=no
    allow=ulaw
    Incoming settings: Leave this entire section blank

    Register:
    Code:
    763NPXXXXX:as3cr37@sip.voipwelcome.com:5060/763NPXXXXX
    Then setup an inbound dial rule to match your DID.

    Note: If you're behind a NAT device, make sure to set NAT=yes, define a port range in rtp.conf, setup externip or externhost + localnet in sip_general_custom.conf to define your external and internal IP ranges, and of course forward UDP port 5060 + your UDP RTP port range on your public IP-facing NAT device to point in at your * box.

  3. #3
    Join Date
    Jun 2009
    Posts
    8

    Default Re: Trixbox Asterisk Help

    Quote Originally Posted by SpaethCo View Post
    I had no problems at all gettting * talking to the BYOD server -- the only time I plugged in the shipped adapter was when I initially got it to be able to get my SIP credentials, and then for some testing for a couple days.

    Here's the FreePBX trunk config I used:

    Dial rules:
    Code:
    911
    1NXXNXXXXXX
    1+NXXNXXXXXX
    1763+NXXXXXX
    Outgoing:
    Code:
    username=763NPXXXXX
    type=peer
    session-timers=accept
    session-refresher=uac
    secret=as3cr37
    rfc2833compensate=yes
    qualify=5000
    nat=no
    insecure=port,invite
    host=sip.voipwelcome.com
    dtmfmode=auto
    disallow=all
    context=from-trunk
    canreinvite=no
    allow=ulaw
    Incoming settings: Leave this entire section blank

    Register:
    Code:
    763NPXXXXX:as3cr37@sip.voipwelcome.com:5060/763NPXXXXX
    Then setup an inbound dial rule to match your DID.

    Note: If you're behind a NAT device, make sure to set NAT=yes, define a port range in rtp.conf, setup externip or externhost + localnet in sip_general_custom.conf to define your external and internal IP ranges, and of course forward UDP port 5060 + your UDP RTP port range on your public IP-facing NAT device to point in at your * box.
    This worked perfectly. Thanks for the help!

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