• Setting up an ARD (Automatic Ring Down) Cisco Unified Communication 8.x, Cisco Gateway Router, Speakerbus Turret

    by  • May 7, 2013 • Informational • 1 Comment

    There a quite a few steps involved in setting up ARD circuits for a Cisco and Speakerbus deployment. I will outline the steps that are necessary to get this working in most environments. Some things may differ in your environment, and things may have to be tweaked slightly to get it working correctly.

    I’ve provided a simple diagram to help you visualize how things are set up in our environment:

    Voice Diagram Cisco Call Manager and Speakerbus

    I’ll assume you already have your Speakerbus Turrets registered to your CallManger. If you don’t just make sure you upload all of the Speakerbus .cop files, and create a device profile for the unit before attempting to register. Hardware and software and Telco in our environment:

    Our configuration

    T1 Line Provided by IPC (we have more than one line for ringdowns but for this article we only need to provide configuration details for one) Cisco 2921 Voice gateway router running .H323 for communications with CallManager (or Cisco Unified Communications Manager CUCM) and a T1/E1 slot terminating our circuit from IPC  (Version 15.2(x)) Cisco Unified Communications Manager 8.6 Speakerbus iManager Speakerbus iD808 turret.

    Steps that are involved:

    1. Configure your gateway router to communicate with CallManager via the H323 protocol. 2. Setup your PLAR line on Cisco CallManager 3. Setup the Virtual Private Wire on Speakerbus iManager

    Configure the H323 Gateway

     

    Cisco CallManager

    1. Create the H323 Gateway by browsing to Device -> Gateway -> Add New.
      • Give you device a name (I use the IP address or the name of the router)

       

    2. Give it a description that will help other people understand what it is used for.
    3. Select your Device Pool (default in most cases)
    4. Call Classification (default in most cases)
    5. Location (default in most cases)
    6. Use Trusted Relay Point (default in most cases)
    7. Signaling Port (1720)
    8. Check the Media termination point required box.
    9. Check the PSTN access box

     

    I came across an issue with barged calls on the turrets. For some reason when a user attempted to barge in on a shared virtual private wire from a Speakerbus turret, the existing call would drop or the voice quality would suffer terribly. It was fixed by checking the Media Termination Point Required checkbox on the H323 gateway.

    Voice Gateway router

    1. If your router is not configured or maybe partially configured you can copy and paste, however if you are in a production environment and your router is already configured please take some caution in modifying your config. I will try to highlight all of the required parts.
      • voice service voip
         no ip address trusted authenticate
         allow-connections h323 to h323 // make sure to allow h323
         allow-connections h323 to sip
         allow-connections sip to h323
         allow-connections sip to sip
         no supplementary-service h450.2
         no supplementary-service h450.3
         supplementary-service h450.12
         fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
         h323 // don't forget this
         sip
          bind control source-interface <your interface>
          bind media source-interface <your interface>
      • controller T1 0/0/3 <specify your interface> // this is where you break up the t1 into 24 channels.
         cablelength long 0db
         ds0-group 0 timeslots 1 type e&m-immediate-start
         ds0-group 1 timeslots 2 type e&m-immediate-start
         ds0-group 2 timeslots 3 type e&m-immediate-start // this is the port that is being used.
         ds0-group 3 timeslots 4 type e&m-immediate-start
         ds0-group 4 timeslots 5 type e&m-immediate-start
         ds0-group 5 timeslots 6 type e&m-immediate-start
         ds0-group 6 timeslots 7 type e&m-immediate-start
         ds0-group 7 timeslots 8 type e&m-immediate-start
         ds0-group 8 timeslots 9 type e&m-immediate-start
         ds0-group 9 timeslots 10 type e&m-immediate-start
         ds0-group 10 timeslots 11 type e&m-immediate-start
         ds0-group 11 timeslots 12 type e&m-immediate-start
         ds0-group 12 timeslots 13 type e&m-immediate-start
         ds0-group 13 timeslots 14 type e&m-immediate-start
         ds0-group 14 timeslots 15 type e&m-immediate-start
         ds0-group 15 timeslots 16 type e&m-immediate-start
         ds0-group 16 timeslots 17 type e&m-immediate-start
         ds0-group 17 timeslots 18 type e&m-immediate-start
         ds0-group 18 timeslots 19 type e&m-immediate-start
         ds0-group 19 timeslots 20 type e&m-immediate-start
         ds0-group 20 timeslots 21 type e&m-immediate-start
         ds0-group 21 timeslots 22 type e&m-immediate-start
         ds0-group 22 timeslots 23 type e&m-immediate-start
         ds0-group 23 timeslots 24 type e&m-immediate-start
      • voice-port 0/0/3:2 // this is the voice port that gets created once you channelize the t1
         auto-cut-through // you may or may not need this. It doesn't hurt anything to have it.
         define Tx-bits idle 1111 
         define Tx-bits seize 0000
         define Rx-bits idle 1111
         define Rx-bits seize 0000
         timeouts wait-release 3
         timing hookflash-in 150
         connection plar 1083 // The DN we created for the phones
         station-id number 1033 // The DN we created for the route pattern
      • dial-peer voice 1033 pots // This maps the port to the DN
         destination-pattern 1033
         port 0/0/3:2
      • dial-peer voice 1 voip
         description INBOUND DP from CUCM
         incoming called-number ^10..$ // I used wildcards because I have a bunch of ringdowns with 10..
         dtmf-relay h245-alphanumeric
         codec g711ulaw
         no vad
        !
        dial-peer voice 2 voip
         description OUTBOUND to CUCM
         destination-pattern ^10..$ // I used wildcards because I have a bunch of ringdowns with 10..
         session target ipv4:x.x.x.x // ip address of your call manager
         dtmf-relay h245-alphanumeric
         codec g711ulaw
         no vad
    2. Notes:
      1. You can create individual dial peers should you need to. My example just eliminates the need to create a single entry for every ringdown that I have.



    Setting Up an External Plar (Cisco CallManager)

    1. Setup an external PLAR line (Private Line Automatic Ringdown – Cisco Terminology) that both the Cisco and iTurret phones can share.
      • First you start off by creating a Route Partition. This can be done under Call Routing -> Class of Control ->Partition. Give it a descriptive name so you can identify it when needed. I.E. “PLAR_BRKR1” Try to keep it as short as possible as there is a known limitation per Cisco. http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_1_2/ccmcfg/b03csspc.html#wpxref59325
      • 2013-05-06_14-25-32
      • Secondly create the Calling Search Space. This can be done under Call Routing -> Class of Control -> Calling Search Space. I give it the same name as the Route Partition. “PLAR_BRKR1_CSS“. Make sure to select the Route Partition you just created.
      • 2013-05-06_14-29-47
    2. Create a null or blank translation pattern. You will be using the Route Partition and Calling Search Space you just created in the previous Step. To create the blank translation pattern, browse to Call Routing -> Translation Pattern. When creating the translation pattern, leave the translation pattern blank, select the Partition and Calling Search Space you created in the previous step. One of the most important steps to remember is assigning the DN that will be configured on your gateway router. In my case I assigned PLAR_UBS a DN of 1033. When we get to the router configuration step this is the DN that will be used.
      • 2013-05-06_14-41-36
    3. Now create a route pattern for the DN you just assigned to the Route Translation. In our case it was 1033.  This can be done under Call Routing -> Route/Hunt -> Route Pattern -> Add New. Set your route pattern, partition, and gateway route list. Make sure “Route this Pattern is selected”.
      • 2013-05-06_14-50-14
    4. Now create a DN (Directory Number) that will be used to assign to the phones and turrets. Under Call Routing-> Directory Number-> Add New. I used 1083 as the DN number for informational purposes. Assign the directory number, select the Route Partition and make sure to set the Calling Search Space to the values you just created.
      • 2013-05-06_14-58-29
    5. This part of the configuration is now complete. Once the DN has been created you can assign it to any turret. Things to note when assigning the PLAR: Make sure the busy trigger is set to 1. If you want to assign it to a cisco sip phone button or side car there are some additional steps involved. I will be writing a separate article about this.
      • 2013-05-07_07-58-41

    Setting up the Virtual Private Wire in Speakerbus’ iManager console

    1. Create the Virtual Private Wire. Under Call Servers ->PBX Appearances ->New. Select PBX (you should have already defined your Cisco CallManager PBX) and type VPW (for Virtual Private Wire). The rest of the details are listed in the picture down below. Note that the Address and Destination address should match the DNs you created in the previous steps.
      • 2013-05-07_07-29-46
    2. Once you’ve created the VPW you can assign it to a device. Under users highlight the user you want to edit and select iTurret Layout to manage the phone template.
      • 2013-05-07_07-47-09

     

    If you have questions or comments, feel free to post them below, or if you need assistance getting some of this set up, I’m happy to do so at the rate of $125 an hour.

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    One Response to Setting up an ARD (Automatic Ring Down) Cisco Unified Communication 8.x, Cisco Gateway Router, Speakerbus Turret

    1. Pingback: Adding a PLAR (automatic ring down) to Cisco SIP Phone with SideCar (Key Expansion Module) | VoIPLing.com

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