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  • SIP Trunks ST8 --> FreePBX

    I am a new admin to ST. I am also lost as a fart in a dust storm.

    I am trying to setup 2 SIP trunks to a FreePBX (PBX IN A FLASH) system....

    I've dove into these posts and configured according to MiJaMu's post.....

    Problem is... I don't know enough about either system to get them to talk.....

    I am looking for step by step.

    What I have done so far.....

    ST8
    Create SIP trunk Group
    Assign SIP trunk (2) to Trunk Group.
    Point the individual SIP trunk IP's to my FreePBX box.
    Switch setup ports for one SIP Proxy the other for SIP Trunks
    Sites config'd SIP uncheck the RTF limitation

    FreePBX
    Create one extension
    Create on SIP trunk with info from MiJaMu's post.


    FreePBX says WARNING: This trunk is not used by any routes!

    What steps do I need to configure FreePBX and ST8 to get them to talk to each other?

    off system extension on ST8 but I don't know how....

    what kind of route do I need for FreePBX? :confused1:

  • #2
    41 Views and No comments.....

    concise & quick answer is all I need. I am versed enough to trench the details....


    Maybe one day I can return the same.

    Thanks

    Comment


    • #3
      On PBX in a Flash you need to create both an inbound route and an outbound using FreePBX in order to make and receive calls to and from ShoreTel. Do a Google search for PIAF without Tears. This is a good tool for learning PBX in a flash.

      On the ShoreTel side make sure that you follow Jlorenz's instructions on page 2 of this thread to implement SIP on ShoreTel.

      I've currently got ShoreTel talking to a PIAF server via SIP. You can PM me with any questions.

      Comment


      • #4
        OLDPHONEGUY

        THANKS FOR YOUR INPUT. I HAVE AN INTEGRATED FREEPBX -> ST 8.1 SYSTEM NOW!

        I CAN DO OUTBOUND AND INBOUND DID's along with a conferencing bridge and fax to email.

        I AM STILL WORKING ON INTERNAL EXT... SHORETEL TO FREEPBX IS OK BUT FREEPBX TO SHORETEL IS NOT WORKING........BUT I CAN DIAL OUT TO ANY DID FROM FREEPBX.

        (ST HAS A PRI AND FREEPBX USES TIES LINES TO CONNECT TO IT)


        ANY IDEAS WHAT TO CHANGE IN THE TIE LINE CONF FILE ???? (I ALSO HAVE TANDEMUM TRUNKING TRUNKING TURNED ON IN ST)

        Comment

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