Local SIP line from Avaya IP500 to 3CX v8

Discussion in '3CX Phone System - General' started by stober, Jun 21, 2010.

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  1. stober

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    Hi guys,

    Trying to get my 3cx box to talk to my Avaya IP500 r6. I want to be able to :
    1) have Avaya phones use a dial plan to connect over a SIP line to access 3cx and then out my skype gateway
    2) Route calls to my skype gateway through to my IP500

    Details:
    The two devices are on the same LAN, the 3cx is not a domain member (xphome). 3cx version is latest/greatest. 3cx IP: 10.63.0.15. IP500 is r6.08. IP500 IP: 10.63.0.202. no firewall rules for traffic on LAN.

    Implementation:
    3cx
    -- Setup extension with ID: 6000 and PW: 6969

    IP500
    -- create a sip Line in IP500 config, details:
    Sip Line:
    • Line number: 18
      ITSP Domain Name: 10.63.0.15
      ITSP IP: 10.63.0.15
      Layer 4 Protocol: UDP
      Network Topolgy Info: LAN1
      Send Port: 5060
      Registration Required: No
      In Service: Yes
      Use Tel Uri: No
      Check OOS: No
      Call Routing Method: To Header

    SIP URI:
    • Via: 10.63.0.202
      Local URI: Use Authentication Name
      Contact: Use Authentication Name
      Display Name: Use Authentication Name
      Registration: 1:6000
      Incoming Group: 69
      Outgoing Group: 69
      Max Channels: 10

    Creds:
    • Username: 6000
      authentication name: 6000
      password: 6969
      expiry: 60

    Shortcode:
    • 6N;
      Action: Dial
      Number: N"@10.63.0.15"
     
  2. SY

    SY Well-Known Member
    3CX Support

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    Question:
    Why do you try to setup 3CX Phonesystem as "extension" connected to Avaya?

    Thanks
     
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  3. stober

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    Thats what i'm trying now. I am able to get my avaya monitoring software to see SIP communication protocol to coming through but my extensions are not being reached.

    here is some of what my monitor software sees;

    16704178mS SIP Rx: UDP 10.63.0.15:5060 -> 10.63.0.203:5060
    INVITE sip:6001@10.63.0.203:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.63.0.15:5060;branch=z9hG4bK-d8754z-9d29e6677e577162-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:ThreeCX@10.63.0.15:5060>
    To: <sip:6001@10.63.0.203:5060>
    From: "6000"<sip:ThreeCX@10.63.0.203:5060>;tag=3b665218
    Call-ID: ZTZkZTYzNjFhMDEzYTFlZTI4NDMyZTA3YmZhNDExMGU.
    CSeq: 1 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
    Content-Type: application/sdp
    User-Agent: 3CXPhoneSystem 8.0.10708.0
    Content-Length: 224

    v=0
    o=3cxPS 479761268736 486220496897 IN IP4 10.63.0.15
    s=3cxPS Audio call
    c=IN IP4 10.63.0.15
    t=0 0
    m=audio 7002 RTP/AVP 0 101
    c=IN IP4 10.63.0.15
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=sendrecv
    16704681mS SIP Rx: UDP 10.63.0.15:5060 -> 10.63.0.203:5060
    INVITE sip:6001@10.63.0.203:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.63.0.15:5060;branch=z9hG4bK-d8754z-
     
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