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Want caller-id but Avaya cant give me an answer

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silvercas

IS-IT--Management
Mar 31, 2004
83
US
Below is my system information and version number. We want caller id on our pots and t1 lines. I know the pots lines come in on one of the 408's and the t1 over the 100d. What would be needed to enable caller id? What reprograming would have to be done

Thanks


Magix Release 4.0 v7.0

A Slot # 0: CKE5 CPU
Serial Number Not Available
A Slot # 1: 408 GS/LS-MLX
A Serial Number Not Available

A Slot # 2: 408 GS/LS-MLX
A Serial Number Not Available

A Slot # 3: 408 GS/LS-MLX
A Serial Number Not Available

A Slot # 4: 400 GS/LS/TTR
A Serial Number Not Available

A Slot # 5: 012
A Serial Number Not Available

A Slot # 6: 408 GLM-U
A Serial Number Not Available

A Slot # 7: 012
A Serial Number Not Available

A Slot # 8: 012
A Serial Number Not Available

A Slot # 9: 100D
A Serial Number Not Available

A Slot # 10: 012
A Serial Number Not Available

A Slot # 11: 008 MLX-U
A Serial Number Not Available

A Slot # 12: 008 MLX-U
A Serial Number Not Available

A Slot # 13: Not Used
A Serial Number Not Available

A Slot # 14: Not Used
A Serial Number Not Available

A Slot # 15: Not Used
A Serial Number Not Available

A Slot # 16: Not Used
A Serial Number Not Available

A Slot # 17: Not Used
A Serial Number Not Available
 
I'm not quite clear on the question, but let me take a shot.

If you are looking to receive Caller ID info for your incoming calls, you will need to order it from your service provider(s).

Caller ID is available with Loop Start POTs lines and ISDN service.

Caller ID isn't one of the Magix's strong points. Outside of SMDR, there is no real logging feature. If you have an Avaya Merlin Messaging module installed (R4 for sure), the originating telephone number (if available) will be announced before the message is played.

You can program a button to choose if you see the name or telephone number displayed first when call info is passed to the sets.

If it ain't broke, I haven't fixed it yet.
 
The 408 GS/LS MLX modules are not Caller ID capable.

The 408 GLM-U might be, depending on the firmware version.

The 100-D, used with a "plain-vanilla" T1 does not show CID. If used with a PRI, it will show CID Number ONLY, but not the name.
 
Thanks for the replies.

For the pots line I'm under the impression I would have to update to a 408 with id? After I install the new 408 with id do I have to reprogram anything?

Lastly on the T1 we use a paradyne unit would that assist with caller id?
 
The 408 GS/LS-ID/MLX should be able to replace the 408 GS/LS MLX without any additional programming.

There is an optional feature that will allow Caller ID info to appear on the displays before the sets start to ring.

As TTT noted, you will need PRI (ISDN) service in order to get Caller ID numbers delivered to you via the T1. No other variety of T1 service will give you Caller ID with the Merlin Magix.

If it ain't broke, I haven't fixed it yet.
 
Both Dexman and TTT are correct!

The Paradyne unit is most likely a CSU (Channel Service Unit), which is used for DS1 service regardless if the DS1 is configured as CAS (Channel Associated Signaling) or PRIMARY ISDN hi-capacity service. The CSU does nothing to the individual channels. It provides equalization, termination, line build out, and monitor and/or test functions for DS1 service. ISDN protocol is transparent to the CSU. Hence, CID capability is not affected by the use of a CSU.

Hope that helps!

....JIM....
 
Thanks for all the great replies.

What Im referring to as a T1 might actually be a PRI, I have to check this out further.

From reading other threads I was under the impression that I would have to do some reprogramming but it seems just like a board swap. Lastly from reading other threads I beleive I have to set a ring delay.
 
1 thing that has been missing from the discussion is the presence of an auto attendant/voicemail. Does the system have one? If so, what is it and how many rings before the auto attendant intercepts the call?

If it ain't broke, I haven't fixed it yet.
 
we utilize audix. To correct this I set the ring delay
 
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