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Thread: Register SIP Trunk with Quadro using Static Registration

  1. #1

    Question Register SIP Trunk with Quadro using Static Registration


    This question is close to a previous post here by another forum member, but I am starting a new thread because that post does not actually point me in the direction I need to go. The post is referenced here simply because it implies the Quadro does have the ability to register static SIP trunks.

    So, question is, how/where do we program the SIP trunk(s) from ITSP on the Quadro2x2 when the SIP Registration method is Static IP, and not user name / password?

    Prior, all trunks have been dynamic and we just created extension(s) in the extensions table. When defining those extension(s), we also entered the username and password data assigned to the trunk(s) and checked that it is to register on the SIP server. The ITSP then validated that data and allowed the trunk to register. I do not see any place where I would indicate username/password vs. static in the set up anywhere. Since the extension manager expects a user name and password, how do you get past the editor forcing a user name and password?

    Am I totally off-base here and not understanding something? I have searched the Admin Guide for the Quadro multiple times and just do not see this mentioned anywhere.

    BTW, we do not have to register static, but would prefer to do so just for added security.

    Any input would be greatly appreciated.

  2. #2


    Re - So, question is, how/where do we program the SIP trunk(s) from ITSP on the Quadro2x2 when the SIP Registration method is Static IP, and not user name / password?

    Under the Call Routing --> call Routing Tables --> make a route with what ever you want to preface the route by and send the call to an IP Trunk.

    Ie - IP Trunk - Digits - 10*? -- NDS 2 -- IP Trunk -- Registrar SIP 5060 U: username P: password -- save and voilah - because the route is available on the CRT then the extensions are able to dial through that route providing it is at the top.

    Hopefullt that gives you enough to work with.


  3. #3


    Hi Kev,

    Thank you for the tip! I do believe that would work perfectly. Of course, I was not able to actually place a call using this method because the trunk credentials being used to test this with are for dynamic IP and not for static IP. The reason I believe your tip is accurate though is because I checked the logs and see that the call failed due to invalid SIP trunk access method. If the trunk being tested with was set up with ITSP as static and not dynamic, this would have most likely worked.

    Here is an excerpt of the log, with the SIP addresses masked out of course:
    SIP/2.0 403 Forbidden
    Via: SIP/2.0/UDP *.*.*.*:5060;rport=5060;branch=z9hG4bKEPSVBUS4d4c1 490-9527-4c51-bb97-c4442dfb23f8
    Max-Forwards: 70
    From: "x203" <sip:***********@******************.net>;tag=14108 78146edb84bac-f9de-4f3b-a00d-d0278466e763
    To: <sip:********@*****************.net>;tag=tBQ859H2Q H4UK
    Call-ID: ********@**************
    CSeq: 381 INVITE
    User-Agent: Broadvox Fusion
    Accept: application/sdp
    Supported: timer, path, replaces
    Allow-Events: talk, hold, conference, refer
    Reason: Q.850;cause=21;text="CALL_REJECTED"
    Content-Length: 0
    X-Disconnect-Reason: Source IP Not Mapped to Trunk
    Once I have the static trunk turned up, I will test again and update this thread.

    Thank you for taking the time to help all of us out!

    Last edited by tchancev; 09-16-2014 at 01:55 PM.

  4. #4


    Very old thread but I just realized I never followed up on it.
    Kevin, you were correct in the setup.
    All we had to do was create an inbound route using specific 10-digit inbound #, discard all 10 digits, prefix internal PBX extension like '00', or in our case a ring group extension #, we entered the SIP server address and 5060 port, and left user name and password empty since there is no username and password attached to the static IP trunk.
    Calls coming in to that DID routed perfectly over the static IP SIP trunk -- no registration required.
    For other inbound DIDs, we just created similar inbound routes.

    For outbound, much the same as inbound with just different source dialing pattern and discards/prefixes but then on out through the same static IP SIP trunk.

    So, thank you again for all the valuable help here....

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