Announcement

Collapse
No announcement yet.
X
 
  • Filter
  • Time
  • Show
Clear All
new posts

  • FreePBX to shoretel SIP

    I have been trying all day to get a freePBX server to connect and place outbound calls on the shoretel trunks.

    Here is the plan:
    setup a SIP extension on the shortel 9.1 system, setup a trunk in freepbx to register using that extension. make outbound calls using said SIP connection to shoretel box.

    I am using X-Lite to connect to freepbx, dialing extensions on the freepbx works with X-Lite, which means it is connected to the freepbx.

    FreePBX:
    Outgoing settings:
    host=192.168.70.5
    username=outboundIVR1
    secret=12345678
    type=peer

    Incoming Settings
    Usercontext: outboundIVR1
    secret=12345678
    type=user
    context=from-trunk

    Registration:
    outboundIVR1:[email protected]

    shoretel user setup:
    FirstName: outboundIVR1
    Number: 8969
    Client User ID: outboundIVR1
    SIP Password: 12345678

    Like i said, it registers with the shoretel system, but the minute i try to place a call from the freePBX system, i get a busy signal.

    In shoretel here is the registration info:
    Name: SIP-CE7B1FD5D8C89346A5599335B797F354
    Site: Corporate
    Switch: HQSG90
    MAC Address: CE7B1F97F354
    IP Address: 192.168.20.111:5060
    Current User: outboundIVR1
    Home User:
    Phone Type: SIP
    Assign To: < None >
    Button Box Order: < None >
    Credential DN: 8969
    User Name: outboundIVR1
    Contact: sip:[email protected]
    User Agent: Asterisk PBX 1.6.2.13
    Expiration: 9/30/2010 6:50:28 PM
    Address of Record: sip[email protected]

    Any help would be appreciated!

  • #2
    where is your dialplan?

    Comment


    • #3
      My recommendation would be to make them tie trunks between the two rather than having the ShoreTel end be a sip extension. As a sip extension you will run into lots of problems trying to get it to pass calls out and the like, trunks are better suited for this application.

      Comment


      • #4
        I have sip trunks to my asterisk box, which is running mostly as a conference bridge, but for our Vosky, I'm also using it for DISA.

        For calls back, I have

        exten => _8XXX,1,Dial(SIP/shoresip/${EXTEN},40)

        shoresip being the sip profile I am using across the sip trunk to shoretel.

        Comment


        • #5
          Thanks for taking the time to reply.

          I was trying to avoid having to pay for a SIP trunk license(we are non-profit), i had a few extra SIP licenses from the original install.

          Here is my dial plan, i just modified the original 0_9 plan:

          1NXXNXXXXXX
          NXXNXXXXXX
          XXXX

          Then the trunk sequence is: SIP/APC_SIP

          Note: I did try to create the SIP trunk, but i think the directions on these forums are leaving things out. If trunking is the only way, would you mind commenting on the previous instructions posted on these forums?

          Here is the link i was looking at:
          http://www.shoretelforums.com/forums...k-trixbox.html

          Another reference:
          http://www.shoretelforums.com/forums...sip-trunk.html
          Last edited by Icetoad; 10-04-2010, 10:58 AM.

          Comment


          • #6
            This is what the config file says:

            [from-trunk-sip-APC_SIP]
            include => from-trunk-sip-APC_SIP-custom
            exten => _.,1,Set(GROUP()=OUT_2)
            exten => _.,n,Goto(from-trunk,${EXTEN},1)

            Comment


            • #7
              Is your shoretel SIP profile setup a specific way? I only have two profiles, _system and one for our conference phones.

              Comment


              • #8
                I didnt mention this, but i was able to get the X-Lite client to connect and make calls with the shoretel system.

                I was also able to get the same X-Lite client to connect to the asterisk box and make calls to the other extensions....

                Is there a way to mimic the way X-Lite talks to shoretel on the FreePBX install? Then i could use the sip lines.

                Comment


                • #9
                  ok, i was able to get outbound calls working from the asterisk box. I still cannot call asterisk extensions from the shoretel system, they go straight to asterisk voicemail. I also cant call shoretel extensions from asterisk.

                  However, the main goal was to use SIP users to call outbound to home phones. We are going to be doing appointment reminders by phone.

                  Here is the asterisk configuration:
                  Trunk Description: 8969
                  Outbound CallerID: 8969
                  Outbound Dial Prefix: 9
                  Trunk Name: ShoreTel

                  Peer Details:
                  disallow=all
                  allow=ulaw
                  canreinvite=no
                  dtmfmode=rfc2833
                  fromdomain=192.168.70.5
                  fromuser=8969
                  host=192.168.70.5
                  insecure=very
                  secret=12345678
                  type=peer
                  username=outboundIVR1
                  user=8969
                  authname=outboundIVR1
                  qualify=yes
                  dtmf=auto

                  User Context: 8969
                  User Details:
                  context=from-trunk
                  host=192.168.70.5
                  secret=12345678
                  type=user
                  username=outboundIVR1
                  disallow=all
                  allow=ulaw

                  Registration String:
                  8969:12345678[email protected]/8969

                  On the shoretel side, we just setup a user with a sip password. the krux of getting the sip to work was the registration string. The the trunk configuration.

                  Still dont have the extension to extension, if anyone has any ideas, that would be great.

                  BTW: in asterisk, you can run "show sip registry", which will let you know if its registered or rejected. Lots of log file parsing to figure this out.

                  Comment


                  • #10
                    Welp, outbound calling works with this config, but now it seems the automated teleyapper cant determine if the person at the other end picked up. So if they let it ring more than once, the audio is already in motion when the called party picks up.

                    Is there a setting i missed in shoretel that prevents the shoretel system from seeming to answer the call before its placed to the called party? It must be related to the fact that it hits a trunk then goes outbound.

                    Comment

                    Working...
                    X