Replacing PC Console with regular set

[icartmanxx]

Our PC Console system crashed on us and no matter what I try, I cannot seem to bring it back online. To keep us operational I swapped in a 6424D+ set with the expansion buttons. We have been running this way for a few weeks and our receptionist now prefers the regular telephone to the PC software.

The programming on our system (which I inherited) had the following on page 3 of the "ch trunk 1" page. It took Called number 1000 (last 4 digits of our main number) and Inserted 1001 (extension of our PC console). To get the new set working I changed this page to Insert 1000 when 1000 was called. Probably not the right way to do this, but it got the calls routed to the new telephone and our receptionist could answer.

What we are noticing now is if a caller dials an extension and tries to hit 0 to reach reception a message comes on stating that transfer cannot be completed. I assume because the PC console is offline and it doesn't know what to do.

How do I get the operator (number 0) to route properly to extension 1000 now?

Suggestions would be appreciated...

 

[Switchman]

Hi,

I had to do this a while back, I had to change the dial plan table so that first digit 0 was an extendion rather than attd.

You can then either add a vdn 0 and use a vector to route it or just create an station 0 and divert the calls to 1000 with a coverage path.

Let me know if you need a hand doing this.

- Dave

 

[icartmanxx]

Thanks for the suggestion Dave. Some interesting things came out of this...but unfortunately I still cannot complete the calls.

When I looked there was already extension 0 defined with cov path 25. I checked cov path 25 and it was set to 1001. Aha I thought...I changed it to extension 1000, but still get the same error...

"Transfer cannot be completed, try again later"

I checked my extension and the cov path for it is 1. I checked cov path 1 and it has Point1 = "h1" and Point2 = "attd". I tried changing Point2 = "1000", but I still get the same message listed above.

Certainly good suggestions, and I thought you had it solved for me...but it still doesn't want to work for me.

Any other thoughts??

 

[Switchman]

Another long shot,

Have you tried renumbering extension 1000 to extension 0?

 

[icartmanxx]

Just revisiting this problem. I noticed today that if I dial "0" from my telephone set it rings through to our new reception telephone (extension 1000) properly. It would therefore appear that the routing of calls works properly, but the transfer from voicemail doesn't work. I still get the message "Transfer cannot be completed. Please try again later".

Any other thoughts on how I can get this working without replacing the attendant phone?

 

[vstarrosie]

what do you have for Voicemail? I'm sure there's something in your voicemail that indicates your 0 out option. Change that to the correct location and retest. I'm sure it will be corrected for you.

 

[icartmanxx]

We're using Audix for our voicemail. There doesn't appear to be any options for it in terms of the 0 out option. From my understanding the coverage path within Definity would control where calls get routed. As I mentioned, it was set to "attd" and I changed it to both "1000" and "0", but neither seems to have an effect.

 

[icartmanxx]

OK....have a spent a bit more time on this today. I'm trying to map out what happens when a call comes in to get a better understanding of where the problem lies. Below is my understanding of the current process, please correct me if I'm wrong.

1. Inbound call directly to users extension.
2. If not answered, Cov Path 1 takes effect
3. Cov Path 1 has "h1" in Point1 and "1000" in Point2.
4. As long as h1 is not busy, the call will be routed to it.
5. h1 is configured as group extension 8000. The group member assignments are 8001, 8002, 8003 and 8004.
6. Extensions 8001 - 8004 are defined as extensions for Audix and each is assigned a port on the switch. There is no Cov Path defined for these extensions.

What is happening is I call in and get routed to Audix (if line isn't picked up). To me this indicates the call is being routed through steps 1 to 6 above and my call is passed to one of the extensions 8001 - 8004 which put me into Audix. Does this sound correct?

I still can't figure out where I define what happens when I press "0" on my phone while I'm in Audix. As I said, I believe I'm all the way through my mapping above and in the Audix system. When I push "0" though I get the message "Transfer cannot be completed. Please try again later".

I believe I have a better understanding of where I am in the system now...and I would imagine it's a dead easy fix...but for the life of me I can't figure out where I set the "0" opt out option in Audix.

Any thoughts on this would be appreciated...

 

[mojoputter]

You make the change in Audix. When your in audix, type change sys feature, what you want to change is, CALL TRANSFER OUT OF AUDIX, go to the 3rd page. Change it to enhanced_no_cover_0, and then change the covering extension to 1000.

 

[icartmanxx]

Mojoputter...you the man!!

Thank you, thank you for finally saving me from this issue.

Your post definitely fixed the problem for me. Only thing is I couldn't select the "enhanced_no_cover_0" option and had to leave it as "basic". Does this present a problem? When I tried the enhanced option I got words to the effect that it was not a valid option.

 

[mojoputter]

You only have basic because of the integration of the Audix to the pbx. You can also set that up in each subscriber's mailbox, on the first page of change subscriber xxxx, there is a covering extension field. If it is blank, then the caller will forward to whatever the extension is set in system parameters.