Problem implementing call back script

I'm having problems getting my message call back script to work.  The idea is that someone makes a call and if no agents are available after a few minutes they'll get a prompt asking them to leave their phone number for an agent to call them back.  The bulk of the script is taken from Cisco's script repository.  Everything works except for when the call terminates and is supposed to place a call to the BaseLineMessageCallback.aef script which will seek out the next available agent and then play a message stating the caller's phone number and a brief message.  Seems like no matter what I try when it goes to redirect the call to the BaseLineMessageCallback script the status is BUSY.  Here are the steps that I've done so far
1.) Created trigger for BaseLineMessageCallBack with the phone number of 6580
2.) Created a 2nd Call Control Group with the ID of 5, CTI route points are 7300 and 7301
3.) Created separate Dialog Group with the ID of 1
4.) The AdvMessageQueueing script is using the same CSS and Partition as BaseLineMessageCallBack, phone number is 6507
5.) Set Media Termination Point to True for Call Control Group 5.
6.) AdvMessageQueueing and BaseLineMessageCallBack are using different call control group and dialog group
I'm able to call both the 6580 and 6507 numbers without a problem from my phone.  The problem is just the redirecting from 6507 to 6580.
Any ideas?

There's a lot more that we'll need to know, to properly figure this out for you.  I would suggest this: start out small, and validate that you can even make a call with the Place Call step to a trigger, since that seems to be the area you've highlighted as a problem.
You can do this very easily by creating a brand new script with a single variable and two steps.  It's really just that simple.  No extra trigger, no application, just the one script, with its one variable, and two steps.  You can think of it like a disposible napkin that you'll use one time and then throw away.
Script1
Variables
Contact c = null
Steps
StartPlace Call (dest = "6507"; timeout = 30; CCG ID = 5; dialog group = 1; contact = c)  Successful    /* Place a comment step here, so when the called script answers, we should see this highlight in red. */  ...End
Start a reactive debug in the script editor: CTRL+Shift+R
For the Script, select: AdvMessageQueueing.aef (I think I read your post correctly and 6507 is the AdvMessageQueueing.aef script, right?  If not, let me know because that's important for this test)
For the Timeout, select: 60
Click OK
Now, make sure your Script1 window is the active window and press F10 three times (1 time to start active debugging; 1 time execute the Start step; 1 time to execute the Place Call step - Note 1: the Place Call Step will remain highlighted in red for now - Note 2: If you press F10 too quickly, the editor may not recognize your key press, so be sure to pause between key presses at this juncture).
What you should see happen is, a new window should open, and it should be the AdvMessageQueueing.aef script, but in reactive debug mode.  Since this is a phone call test, you will need to make sure the AdvMessageQueueing.aef script window is active and press F10 until the red highlighted step is whatever step immediately after the Accept step.  This answers the call and back in Script1 you should see the comment step highlighted now.
If you don't see that, then you know there is a problem with a CTI Port in CCG 5 calling a CTI Port in whatever CCG 6507 is associated with.  I know you said the Partitions and CSSes are the same, but I would ask that you just double check that the CSS on CCG5 contains the Partitions for both the 6507 Trigger and the CCG 6507 is associated with.
Let me know if you get past that part, or if we need to address any problems you encounter.
Good luck.
Anthony Holloway
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  • How do I Translation Route a call back into CVP?

    This is ICM 8.5, CVP 8.5 and Avaya CM 5.2.  I'm trying to configure the system to be able to Translation Route a call back into CVP that originally started in CVP.  I've read and read Cisco documentation that talk about various ways it can be done but I think I really need step-by-step instructions to figure out what I've done wrong.  I'm happy to explain it in gory details if anyone wants to help me troubleshoot.  However, I think it would be simpler if someone can point me to a document that outlines a successful configuration and I can find my own mistake.

    Thanks clearbluelou, some of this was very helpful, but I had to set my translation routes up differently. Kris Lambrecht's post was very helpful, but also introduced some confusion, which looking back was very simple...hindsight is always that way right?!?!
    I am going to list out the configurations I used to make this work, most of which are what's listed in the Type
    8 setup in Kris's post starting with Step 7 except its for a Type 10 Network VRU; you don't use Type 8 anywhere, but the configuration is very similar.
    Assumption is that you already have an existing system and are routing calls to CVP already so a lot of the steps in the config guide are already done.
    References below are to "Configuration and Administration Guide for Cisco Unified Customer Voice Portal Release 8.5(1)" starting on page 109, step 7:
    http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/customer_voice_portal/cvp8_5/configuration/guide/ConfigAdminGuide_8.5.pdf
    Here's the details of my implementation:
    ICM 7.2.7
    CVP H323, Type 10
    Avaya PBX connected with a PG also running CTIOS
    Destination is the CVP Servers/CVP PGs (PG1), call transfer is originating from an Avaya PBX connected to ICM via a PG/CTIOS server (PG2).
    1. Build a new Service under destination PG, CVP in this case, as outlined in Step 7 from the CVP Config Guide.
    - Service
    Peripheral Number: 2
    Peripheral Name: Pre_Route
    Name: CVP_PG1.Pre_Route
    - Route
    Name: CVP_PG1.Pre_Route
    -Peripheral Target
    Not required
    Labels
    Not Required
    2. Build Network Trunk Groups, as outlined in Step 8 from the CVP Config Guide:
    - Network Trunk Group
    Name: CVP1_NTG
    Peripheral: CVP_PG1
    PeripheralNumber: 200
    PeripheralName: CVP1_NTG
    Name: CVP_PG1.CVP_NTG
    Trunk Count: Use Trunk Data
    3. Build Translation Routes using the TransRoute wizard, I built a pool of 10 DNIS 1000-1010; these are dummy DNIS; outlined in Step 9 from CVP Config Guide; here's what they should look like in
    the TransRoute explorer when finished:
    -Translation Route
    Name: TR2CVP1
    Type: DNIS
    -Route
    Name: CVP1.1 - CVP1.10
    Service Name: CVP_PG1.Pre_Route
    -Peripheral Target
    DNIS's: 1000-1010
    NTG: CVP1_NTG 
    -Label
    Routing Client: PBX_PG2
    Labels: 1000-1010
    Label Type: Normal
    Customer: None
    4. Configure Trans Route DNs 1000-1010:
    - Have to be built on the Avaya pointed to a VOIP gateway with Session Target equal to the CVP Call Server (or GK).
    - Add Translation Route DNISs in CVP Call Server Configuration under ICM Tab.   
    !!!Make sure you "Save and Deploy"!!!---->If you don't calls will show up on the ICM as new calls on the Translation 
    Route DNIS's under your CVP Peripheral.
    5. Build Dialed Numbers in ICM:
    -Extension 2000 is the dialed number Agents on the Avaya transfer to.
    -VDN 2000 on Avaya PBX should be routed to an Adjunct Route step which will send a route request to PG2 and kick off a script in ICM.
    -You have to build a dialed number in ICM Dialed Number Explorer under the PBX peripheral (PG2 in my example) PG2.2000, and point it to a CallType and subsequently a script, for example "Avaya2CVP".
    6. Build your ICM Call Routing Script: (follow's ClearBlueLou's description)
    - Route the call through typical CVP call setup initialization nodes
    -Route to Trans Route to VRU node with "CVP_PG1.Pre_Route" as the service selected, and "TR2CVP1" Trans Routes 
    selected.
    -Then send to a microapp.
    Your test call should have exited the success connection of the TrasRoute to VRU node and will continue routing with all call variable information will route (and show up in TCDs and RCDs) under Dialed Number PG2.2000.

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