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How To Create First Outgoing Call Using Star Communications

Tom Cooper
2017-04-09
2017-04-09
  • Tom Cooper

    Tom Cooper - 2017-04-09

    I've succesfully installed & set up two SIP devices & can call & talk fine between each. I've also ran an update via the "raspbx-upgrade" command in Terminal - so there appears to be no issue with connecting the the internet.

    However, when I try to call out using a toll free number & the Star Communications outgoing SIP trunk, I hear the message "Your call cannot be completed as dialed. Please check the number & try again.".

    To confirm my actions within the FreePBX control panel so far:

    1.Settings > Asterisk Sip Settings > Detect Network Settings In NAT Settings
    2.Applications > Extensions > Created 2 Chan_SIP Extensions - configuring only the User Extension & Display Name Fields & copying the secret.
    3.Inserting the SIP Extensions into the SIP Phones
    4.Dialling American Airlines Toll Free number "800-433-7300" in three different ways (I'm not from the US so unsure of the correct number):
    - 8004337300
    - 18004337300
    - 0018004337300

    Do I need to do anything else or should it work?

     
  • Gernot

    Gernot - 2017-04-10

    You need to call the number this way:

    +18004337300
    

    The default trunk from the base image requires the +1 prefix, because it has to be compatible with all the numbering formats around the world. If you live in the US you could modify the outbound route such that it also accepts 18004337300 or 8004337300.

     
  • Abdenour Bouras

    Abdenour Bouras - 2017-08-18

    Hi

    I've followed step by step guide and i've configured Star communication sip trunk on my raspbx. Unfortunately can't get anything (call failed). I've done port-forwarding on my openwrt firmwares router (Tomato v1.28 Beta), didn't get anything and i tried with putting my raspbx host on DMZ but no use.
    

    Any help please ?

     
  • Gernot

    Gernot - 2017-09-18

    I've tested the Star communication trunk again with the latest beta image dated 18-09-2017 and it works fine. I had to configure the public IP in Asterisk SIP settings and I also forwarded SIP and RTP ports 5060 and 10000-20000 on my router. Other than that no further action was required.

     

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