Startalk AA problem

[Bower]

I have a bugger of a problem I was hoping someone could help me get insight too. I have a site with a Norstar CICS running DR2.0 and a Startalk Flash running 1.7.00. At this site I have an AutoAttendant set up because they really don't have a full time recepotionist at that office and the AA fills that role beautifully.

The problem is one of our clients can't dial any extensions when they reach the AA. They can dial # to dial by name and it works but if they dial any of the extensions, they recive the message "You have dialed an invalid extension". I have gone to the client's site and tested it out on other Norstars we have with AAs and it works fine. We have had the CO come to the client's site to test their phones and everything checks out ok. The CO tech has also put a number grabber between the Key system and the CO line and we have had the client call and the numbers are coming across correctly. Lastly we have replaced the Startalk with a new one just to see if that was the culprit and the problem persists.

Our next step seems to be to replace the whole CICS but that seems like a shot in the dark to me as I can't see how the CICS would be the offending equipment here. Any insight you guys might have would be appricated.

Bower

 

[ajntek]

Can they call these extensions internally?

ajNtek

 

[Bower]

Yes, no one else internally or externally has problems dialing extensions. Just this one client.

Bower

 

[thtek]

If he is dialing from a his private line (if he has one), did you check to see if it was set to Pulse or Tone in "Lines" for the dial mode?

What does the phone display indicate when the ext number is dialed?

Has this always been a problem or only after the AA was set up?

Have you swapped station ports to see if the trouble follows?

What ext # is having the problem?


thtek

 

[Bower]

They are dialing from a private home. All phones in the house are set to tone.

The one phone with a display shows the correct digits dialed.

The trouble is for all extensions on the KSU from that POTS line only. Mind you dial by name still works COMPLETELY i.e. the voicemail is getting the correct digits when the user dials by name.

 

[AlquestTech]

They must be a very important client if you would change your whole system for them.

Various Auto Attendants have different tolerences on the DTMF. Also not all phones generate "good" DTMF (I.O.W. - there is a fair amount of 'slop' allowed.

However some automated systems do not carefor the slop and will not recognize the tones as valid.

The digit grabber you used, did it also display the frequencies or just the digit.

Digital sampling rates can further muddy up the waters. I know we have one automated vendor (FedEx) that will not recognize our digits as valid when sent through our PRI (T-1) Even though I monitored the tomnes and they sounded good and the numbers were ok, the frequencies, (I am assuming ue to the sampling) had crept just toward the edge of tolerance. Everybody else recognized them, but not Fed Ex's Automated system.

I fixed this by just matching that number in the call routing table and sending it out via pots instead.

Perhaps your client is in a similar position and could try to impliment a similar solution?


(Here is a good starting place -

http://www.hut.fi/~then/mytexts/dtmf_generation.html)

 

[MarvO1]

You most likely need to adjust the DTMF tone duration parameter.
The minimum industry specification for DTMF tone is 40 ms.
Some phones can send a shorter tone.
Log into voicemail admin as normal, then press 8
When it asks for a password, use MAINT (62468). Make a note of what the current setting is before changing, because if you adjust the duration too far, you will start getting a new problem called Talk Off. Usually female voices being confused with DTMF. As you improve one, the other gets worse.
Remember, 40 ms is the indusrty standard.



MarvO said it

 

[thtek]

Let us know if that was the problem.
Thanks

Thtek

 

[Bower]

I managed to go to the site and check out the values but every value besides 40 I tried to enter came up invalid. Can you guys tell me the syntax I need to enter into this field? Thanks for your help.

Bower

 

[Bower]

I found it. Only options on the Startalk seems to be 40, 60 or 80. I will test 60ms and see if that helps. Thanks.

Bower

 

[Bower]

Well, I've tried 60 and 80ms settings on the DTMF and the problem still is there. Was a cool thought though.

Bower

 

[Arr]

Bower,
Perhaps try this...

At the far site (without the voice mail) take your butt set or a NT9317 set or similar and bridge the CICS so that you dial straight out on the CO trunk which the CICS does.

Also try using the 'long tones' feature through the CICS. IF long tones work, it is possible the CICS is generating a very short DTMF signal which that particular VM, because of line conditions or whatever doesn't like. Marv0's suggestion of changing the recognition time in the voice mail is a good one BUT you then run the risks he(she?) outlined while the real problem may be in the CICS.
Long Tones is F808.

Cheers
PhM