SIP Call routing

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  • contict2007
    Member
    • Dec 2014
    • 3

    SIP Call routing

    Hi All,

    I've got a challenge at a customers site with 3 different systems with a IP office in the middle.

    System A: Ascom nursing system
    System B: Avaya IP Office
    System C: Broadsoft PBX at HQ

    The Avaya is in the middle and connected with SIP trunks to both other systems. It's not possible to create a SIP trunk between the Ascom system and the Broadsoft PBX.

    Situation I have created is the following:


    System A -------SIP------>System B----------SIP-------->System C


    Following is working correct:

    Call from System A to B = OK
    Call from System B to A = OK
    Call from System B to C = OK
    Call from System C to B = OK

    Call from System C to A = NOT OK!

    SIP URI on the LIne between system B and C is filled with * so system will route icomming calls with a match in the 'Incomming Call Route' table. This table is filled with an empty field 'Incomming Number'. Within the Destination field I use a shortcode to route those calls to the second SIP trunk. (Trunk between system A and B)

    Unfortunatedly the shortcode is not allpied at the right way. So incomming calls are not routed to the second trunk at all.. What am I doing wrong? I see the shortcode is used in the user part of the SIP URI....

    Shortcode: 4xx, dial, 4N, line 17

    When call comes in to extension I see to Header: [email protected]

    Outgoing SIP URI to line 17 is: [email protected] (IMHO this must be [email protected])

    Hope you can help me folks!

    Many thanks in advance.
    Tim
  • zakabog
    Genius
    • Aug 2014
    • 300

    #2
    Do you have an incoming call route for 4XX with a destination of 4# setup (it will throw an error but ignore that and give it a go)? Try that and see if it works, otherwise you're going to need to setup a call trace and see where the call is getting lost.

    Comment

    • contict2007
      Member
      • Dec 2014
      • 3

      #3
      I've got a working configuration now as it seems!! I changed the way calls are routed by the SIP trunk. I changed it to 'To Header'in stead of Request URI. Now the calls seems to be routed the right way.

      Thanks for the repley!

      Comment

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