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Thread: Trixbox Asterisk Help

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  1. #1
    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.

  2. #2
    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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