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Help no incoming calls - Gamma Telecom

Discussion in '3CX Phone System - General' started by lowbug, Apr 3, 2011.

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  1. lowbug

    Apr 3, 2011
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    I'm new to this :oops: , i have downloaded the latest 3cx free. My network is setup with a win 7 pc ( firewall off ) and a draytek 2820 setup as http://www.3cx.com/blog/voip-howto/draytek-firewall-voip/

    I have setup it and sucessfully got outgoing calls no issues, even got the iphone as an extension. However incoming calls dont even appear in the call log :(

    Its username and password authenication ( not the older gamma platform which is IP auth )
    I got this>

    Dear Customer,

    Your IP Direct Connect Order for Gamma Order Number: ??????? has now been provisioned and is Ready for Acceptance Testing.

    The details of your provisioned order are as follows:

    Extension number: 000xxxxxxxx7

    Password for authentication: yyyyyyyyyyyyyyyyyyyyy
    SIP server: gw3.theiptele.com

    (The unique network identifier of the CPE that is connected to the IP Direct Connect Service)

    Number of Channels: 2

    Allocated DDI number(s):


    (Please note if you require to port numbers to the IP Direct Connect endpoint, this process can only commence once the new IP Direct Connect node has successfully been provisioned and tested)

    The Endpoint xxxxxxxxxxxxxxx, is now in a position to be tested, can you please undertake the pre-defined Service Acceptance Tests that were outlined in the e-mail that Gamma forwarded to you on Order Acceptance. (The tests are included in an attachment to this mail for your information).

    The following information defines the configuration details that relate to the Gamma IP Direct Connect (IPDC) solution. The details should be used in the configuration of the CPE that is planned to be connected to the Gamma Telecom IP Direct Connect service:

    1. Gamma Telecom IPDC SIP gateway: gw3.theiptele.com Please note that IPDC authentication is based on User authentication, the solution does not require the CPE's IP Address.

    2. The SIP header requirements in the INVITE packets originated from the CPE should be set as follows: - SDP payload must be present, - From: header must contain public facing interface, originating CLI in the display name field and SIP username in the userpart e.g.: "From: 'CLI_number' 'SIP_username'@'public_facing_interface_IP_address'>") The SIP username is the same as the extension number mentioned above, - To: header must contain IPDC SIP gateway address or domain and the called number with leading zero or in E164 format.

    3. The CLI numbers are validated against the Gamma assigned geo number ranges for each endpoint and will be accepted in the following formats: - "+44"+area_code+phone_number - "0"+area_code+phone_number - country_code+area_code+phone_number These formats can be presented in: - the PAI (P-Asserted ID) header of the SIP Invite message - the R-PID (Remote Party ID) header of the SIP Invite message - the From header of the SIP Invite message (Please note that we do not support P-Preferred ID headers for CLI presentation.)

    4. The Traffic on the following ports must be forwarded through relevant routers and firewalls on the customer premises to allow access to the Gamma IP DirectConnect Network TCP and UDP 5060 egress/ingress to gw3.theiptele.com UDP all ports between 1024 - 65535 egress/ingress to gw3.theiptele.com

    Any ideas on where it is going wrong? I desperately need it working, i have even used the provider example ( which needed modding to fit the supplied info ) here http://www.3cx.com/blog/voip-howto/3rd-party-supported-voip-providers/?wpmp_tp=1 but no joy!

    Aggghhh!! :cry: :cry:
  2. Discovery Technology

    Apr 19, 2008
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    Have you allowed inbound SIP UDP 5060 from your VoIP provider?

    Ensure that you lock this down when you open this as your system would become vulnerable to hacking from outside sources if SIP is open to all outside IP's though.

    Have you set your SIP trunks to register?

    If your SIP trunk registers with your ISP and you allow inbound SIP (I should also include RTP inbound) then your phones should ring when your VoIP number is called.
    Stop hovering to collapse... Click to collapse... Hover to expand... Click to expand...
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