Merlin Mail VM and AA together

[altaphon]

Thanks to much help here our system is up and in use, but there is still one programming glitch that I can't figure out. It is a Merlin Legend system in key mode, R7 V12, with a 4-port MLM 007. Two ports of the MLM are allocated to AA and a calling group goes to those ports on coverage (we prefer to grab the call before the AA gets it). The AA works fine, callers dial a digit to be transferred to a phone. If that person doesn't answer, the call goes back to the AA. We're trying to change this so that the call goes to the dialed person's VM instead when it isn't answered. We have the remaining two ports assigned to VM, and a different coverage group. The problem is that when we put extensions in that coverage group, incoming calls (which also ring on those extensions) go to the first extension's VM instead of the AA. How do I arrange it so that VM only answers calls that ring on the extension number, rather than all the other lines on that phone?

 

[merlinman]

altaphon - The solution is simple.

First off, set all of your VM Ports for CALL ANSWER, not the 2 for AA as you have it now.

Then, assign the LINES to a SEPARATE Calling Group. (771?)

Next, Program that GROUP to OVERFLOW to your VM Calling group, with the FOLLOWING OPTIONS:

Number Based = 99
Time Based = 20 (That's seconds don't you see.)

Now, when a call comes in on your LINES, if no one answers in 20 seconds (Or what ever you want to set it for) the AA will answer.

AND - - If the caller selects an extension, BINGO, that guy will get the call transferred to him!

Let us know how that works out.....

 

[altaphon]

Merlinman, thanks, but that's not the problem. The AA works fine, a selector code gets the call transferred however I have it set. But, if that person doesn't answer, it goes back to the top of the AA menu and not to that person's VM.

 

[2BAvoice]

Merlinman is right, that IS the problem. You don't have to have ports dedicated to AA for the AA function to work.

 

[altaphon]

If I don't have the port assigned to AA, I get the VM menu, regardless of the class of service I assign. All 4 are now assigned to AA, and now I have this working according to the recipe merlinman suggests... It had been working before, using the somewhat longer route I described. The AA plays, you can when someone transferred to an extension and no one answered; as originally set up, the call went back to the AA menu rather than to the mailbox. I had neglected to set up the calling group for the extensions. Now each extension with a mailbox is listed in coverage group 30 and the mail VM calling group number is the coverage receiver, all works OK. Thanks, all, for your help.

 

[tdaugirdas]

Well - you have it working - but it is not set up correctly. You need to do as Merlinman suggested. Change the VM setting on all ports to call answer. Create an overflow group - lets say 791 - with no members. Create a mailbox for 791 as an AA1 mailbox (type 15 for an 007 MLM). Set 791 to overflow to your voicemail calling group - usually 770. Set number-based overflow to 99, time-based to 20 seconds. Attach the lines meant to go to AA1 to group 791. Now when those lines ring - after 20 seconds - it will flow to autoattendant 1 and play the menu. If you want less rings at the operator - change the seconds to 16. If you want no rings - change it to 1 second - and turn off ringing at the operator phone. If you have multiple AA's in use - pick another unused calling group - say 790 - and do the same - except make it's voicemail box type 16 (for AA2), etc. Attach the lines for the 2nd company - and now calls on those lines will flow to AA2 after 20 seconds.

Tom Daugirdas,
President
STCG, Inc.
stcg.com

 

[altaphon]

Maybe I am missing something, or haven't described the system properly. This is an MLM 007. Tom, my choices for port allocation are all VM, all AA or split half and half. Where is the "call answer" option? Other than that (and I am using 778 instead of 770) I am doing exactly as described and it works fine.

 

[tdaugirdas]

All Voice Mail is what I meant - sorry. With all ports devoted to AA - you will not have voicemail answering inside calls - or calls that may go directly to an extention through a private line that bypasses the AA. All Voice Mail is the default - and there is rarely a need to change it.

Tom Daugirdas,
President
STCG, Inc.
stcg.com

 

[altaphon]

If I set the ports to VM, they answer with the VM prompt rather than the AA menu, regardless of the class of service. Inside calls are answered by VM because I have another coverage group doing this. For some reason, an "all AA" port allocation will allow a mailbox to be defined with VM service, but an "all VM" port allocation will not allow AA service anywhere. Is this a bug?

 

[tdaugirdas]

If it is working as you describe then you do not have your telephone lines assigned correctly - as I described above. How are your telephone lines assigned to voicemail now?? Are they assigned to Group 770? Do you have them assigned via voicemail admin?

Tom Daugirdas,
President
STCG, Inc.
stcg.com

 

[altaphon]

The VM ports (extensions 10 through 13) are assigned to the VM group which is 778, per the MLM manual page 3-4. The CO lines are attached to 771 which overflows to 778 as Merlinman suggests. 778 is of queue type "Integrated VM." In MLM, 778 has class of service 15, which is AA1. The extensions are in coverage group 30, which is covered by calling group 30, per the manual page 4-9.

 

[tdaugirdas]

Whoah...OK - if you want to use convention - here it is. The voicemail calling group (by convention) is 770. You assign as members the voicemail ports. You make the hunt type for this group linear and the group type integrated voicemail.

Then you put all the extentions to be covered by voicemail into Cover Group 30 - I know - you used Group 1.

Then you set group coverage by 770 for cover group 30.

If you want the voicemail to answer immediately - you assign your telephone lines to the voicemail group 770.

If you want a delay - you create a second calling group. You used 771 - and that's fine. This group has no members. You DO NOT change the default hunt type or group type. You simply overflow this group to 770 - and set the appropriate time-based delay.

You then remove your lines assigned to 770 - and assign the phone lines to 771.

Last step is to create a type 15 mailbox for 771 in voicemail administration (assuming you are using Autoattendant 1). You can name it voicemail overflow.

That's it - in a nutshell.

You're finished.

Tom Daugirdas,
President
STCG, Inc.
stcg.com

 

[ttech]

Also check that the transfer recall is set higher to a higher number than cover delay

 

[altaphon]

Yep, thanks, I'm finished. It works. Only problem now is that the TTR doesn't respond well to digits dialed on a cell phone; the timing is too quick and a "4" with a crackle in the middle is interpreted as a "44" which causes the call to go to the operator instead of to the entry for selector code 4. But I guess I can live with that, or dink with the analog line levels.

 

[merlinman]

That is probably a setting on the CELL PHONE.

I fight this all the time.

You maybe able to fix it by having the Cell Phone send LONG TONES when it dials.

Oh, and tell the Cell Phone users to dial VERY SLOWLY.

Or, at least that's what I have to do when I use my Cell to PROGRAM Voice Mails.

 

[dagwoodsystems]

I'm not entirely sure that you can "dink" with the analog line levels. The board's CO ports want a -4.0 db signal at 900 ohms, and it's not adjustable internally.

And while it's possible that a TTR might be at fault, I'd be more inclined to believe that poor radio strength is at fault for the cell phone not senderizing the digits properly. The "crackle in the middle" is coming from the cellular network, not the PBX. I too believe in the promise of mobile communications: Cell phones will be great once they get 'em working.

BTW, why would a "44" send you to the operator and not follow the treatment of Selector Code four?

 

[altaphon]

Now cell phones and digital radio are more in my line of expertise. There are dozens of places where the tone can get corrupted; some systems pass the DTMF audio from the phone, others recreate it somewhere in the switching system, and who knows what kind of lossy audio compression is used to get it to the T1 line out of the cellular domain and into the PSTN. I'm sure the problem is not in the TTR, nevertheless if the CO is delivering -14 or +8 instead of -4, or with a high amount of twist, there might be a problem. This would be fixed with a 2 wire repeater with which I would adjust levels, but the cure would probably be worse than the disease.

I don't know that a 44 was decoded. Whatever it was, isn't in the short list of selector codes that I'm using.

Sorry for the OT ramble, thanks for the help.