Ovislink VoIP 800 SIP configuration

Discussion in 'IP Phones' started by songokumanjimaru, Feb 22, 2011.

  1. songokumanjimaru

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    Hello everyone

    I'm new on this forum, however I hope that anyone may helpme with a Ovislink VoIP 800 that is similar configuration to PLANET VIP XX series.

    I need to use the Ovislink as FXO Gateway, for an elastix PBX, please giveme ideas.

    Kind Regards
     
  2. songokumanjimaru

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    Hello Every one I solved that by myself, after a long hour of work :laugh:

    First Elastix trunk Config:

    Make a SIP trunk with 220 Caller ID and outbound Prefix (220 by my choice to a ovislink dialplan)

    Trunk name "anything"

    ON PEER DETAILS:

    username=admin
    type=peer
    secret=123456
    qualify=yes
    insecure=very
    host=172.16.10.9
    context=from-trunk
    disallow=all
    allow=ulaw
    dtmfmode=Auto
    canreinvite=no

    Save your sip trunk and create an appropiate inbound/outbound routes

    An then magic comes:

    On ovislink VoIP800 need to configure by web console and CLI:

    0. Be sure that your ovislink have a proper SIP firmware
    1. Login to Ovislink web config (user "eitg", pass "123")
    2. On Phone/Hunt Group/Dest. Setting Menu: create a remote ip destination, ID=10 (may be different), Server Address "elastix IP add", ather fields in "blank"
    3. Create two hunt groups, first one ID=10 and destination ID=10 (as remote destination); after create another with ID=20 and set multiple destination ID's as lines (1,2,3,4,5,6,7,8)
    4. Create two phones: One for incoming calls "210" Huntgroup=10, Maxdigits=16, and strip=0, Other for outoging calls "220" hunt group=20, Maxdigits=16, and strip=3.
    5. On system settings, sip parametter settings and outbound proxy settings, type elastix server IP address and port (default 5060)
    6. Also may configure codec settings
    7. At CLI do a telnet coneecion with same user and pass as web interface, CLI is same as Planet gateways.
    8. To make incomming calls type for each port: "set port X dial_in plar 210" when 210 is the phone to poit elastix server
    9. to make dtmf relay (needed to access IVR) do "set copding 6 dtmf_relay on in

    That's All

    I hope that this may help someone, if need information in spanish send me pm.

    Kind regards
     
  3. zfinch

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    Glad you got it worked out by yourself. For future hassle, many high quality Voip Service Providers should have a support team that can help you through these issues, rather then trying to figure it out yourself. But it definitely is nice figuring these things out on your own :)
     

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