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F4 external transfer dialtone sounds like disconnected 4

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TedTheITGuy

IS-IT--Management
Mar 14, 2006
37
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US
I have successfully setup the external transfer to an off-premice office that has a single line phone. F4 +destination code + external phone number. It works but it is not seemless -- it is not professional.

The problem is when the caller dials the internal extension number, the system appropriately grabs a line from the line pool, the caller hears dialtone that sounds like a disconnect, then the external number is dialed by the system and they will be connected if they stay on the line long enough. These callers will be returning calls to a "collection manager". Gernally speaking it is not a call they will be happy to make and they likely would be happy to have an excuse for not returning the call if they can say "I tried your extension as your collection letter states I needed to do but your system just disconnects me" I would like to eliminate excuses and make the system behaive and sound more professional as well. Any ideas MVPs?
Thanks,
TED
 
assuming your voicemail system has CCR, why not try that option
a professional might be able to help you set it up or you can set it up yourself
 
You could also try line redirection.

Sometimes its more helpful to know what it is you'd like to do and why, rather than how you're doing it now.

The more details the better we can assist!

SD
 
Thank you SENK1S and SupportDude for the response.

SupportDude: Let me start by describing in more detail what I want to do.

There is a main telephone number. All callers call into this number and reach an automated attendant with CCR.

SENK1S: just as insight on my background: By now, I have designed, setup ands recorded close to 20 automated attendants with CCR on systems ranging from Nortel, through Lucent, and into Mitel. They are all tailored to the needs of each particular site. Mitel handles the off-site transfers the best but has the most volume loss, just for the record. I have most of my clients today because my predecessors couldn't or wouldn't make the equipment work. They just wanted to sell new equipment. Instead of saying “no” I say “let’s see what we can do”. I have noticed in your previous responses in other posts that you are very quick to suggest professional help. I work for a living and find solutions, therefore I am a professional. Whenever I have sought outside professional help, I find they lack diligence, creativity and usually say it can't be done if they don't find it in the manual. This is what lead me to do it all myself. I am not an expert by any means, I certainly don't know everything but there is always a solution or a work-around if we look for it and ask the right people for new intelligent offerings. I have this current Nortel CCR working pretty well for over 5 years and now the site would like something new added to it.

When callers reach the auto attendant they have about 7 initial options. One of the options is "dial your party's 4 digit extension. What we would like is to have a four digit extension that can be dialed once a caller reaches the automated attendant that will ring at an off-premise analog phone. The desire is to create the impression that the callers have reached the same branch they usually call. But actually have the call seamlessly transferred to and answered off-premise after dialing the four digit extension just as would happen as when they try to reach anyone at the main branch.

I realize there are many ways to accomplish this. Budget and speed of implementation are of course a concern. This extension and it appearing professional is a necessary evil but it will not be a money maker so huge capital investment is not practical for this site. If I have to I will eventually suggest they settle on a less than preferable solution but I hope to find better. The external transfer is perfect except for that brief dial tone that sounds to caller as if the call was disconnected.

MCK offers the 6000 and 4000 solutions via TCP/IP and the MCK 1000 will work over analog phone lines. This is a little costly and too much equipment for the far-end user location.

I can also have the callers reach the voicemail box for the extension at the main branch, then have the voicemail greeting say “press 7 and wait to be connected” (utilizing the “outbound transfer” feature of Norstar NAM voicemail). But then we have just gotten back to HINKY. Hinky needs to be saved for a last resort.

As SupportDude notes, Line redirection is an option. I have set that up before at another site. While I do believe the outgoing call portion of the redirection can use the line pool (which is efficient), I have not figured out how to allow the incoming portion of the line redirection from an automated attendant. I can only do it by dedicating a direct phone line to the extension which is a monthly cost for a phone line that won't be used very often and also a different phone number than the desired main phone line. Supportdude, do you know how to make line redirection work from an automated attendant?

SENK1S? Can you think of a way to setup CCR so it appears the caller is dialing a four digit extension? I thought of trying to setup a menu path based on the numbers of the four digit extension but haven't played with that yet. I suspect the caller would enter the digits too fast and be routed incorrectly.

Ok people, there is my detail, and what I would like. Questions suggestions or comments are always welcome

Thanks again
 
You're right Ted ... there is always a workaround!

Here's a thought.
For line redirection via AA.
You'll need an ASM port or an ATA.
You'll need an analog trunk port.
Send the call (via the 4 digit extension) to the ASM port.
Cross-connect the ASM port to the Analog trunk port.
Assign the appropriate trunk/line to the set.
This should allow you to use line redirect by dialing a 4 digit extension via AA.

If you have a CallPilot 100/150:
You'll need one greeting table and one CCR tree.
Send the call (via the 4 digit extension) to the ASM port.
Cross-connect the ASM port to the Analog trunk port.
Program a Greeting Table to answer the appropriate trunk/line.
Assign a CCR tree to the greeting table and make the home node an external transfer.
This would even allow you to insert a message, if desired, to inform callers to hold while there call is connected.

Sorry for the delayed response ... been really, really busy!

SD
 
SupportDude,

First off, please, no apology required for delayed response. I appreciate your very clever offering in the middle of your being busy. Thanks for reading my long explanations.

I have tried your idea and does work pretty well. I hope some adjustment can alleviate 2 issues.

Setup: ATA port assigned with the desired 4 digit DN (AA calls ATA when the 4 digit extension is dialed). ATA is cross-connected to analog trunk.

I assumed you meant I needed to setup a spare m7310 (using a random extension number) in order to make the appearance of the ATA trunk Line and the redirection. Setup redirect at the spare set from inbound ATA Trunk using line poolA to dial out to the target outside phone number.

In the test environment: dial the 4 digit extension (either from internal intercom or from Auto-Attendant) -- the system says "one moment please", then there is silence for a few seconds (totally okay so far), then ringing, then answered by the redirected target phone number. This was great!! Voice quality and volume are clear, etc., the appearance of professionalism to a caller sounds great, etc.

There are two problems I have encountered so far.

1) Touch-tone signals not being relayed from keypad of initial caller to the far end answering system.

Touch-tones don’t work after the redirect. This means outside caller reaches auto attendant, touch-tones work fine to navigate to the 4-digit extension. But after the redirect, when reaching the far end users voicemail system, the touch-tones do not respond. Even though I didn’t really think this was the problem, I did verify that all phone lines are set for “dialmode -- tone”.

2) Phone lines are remaining tied up after the call is complete.

Following the call, in most of the tests, the ATA will not let go of the line it used from the line pool. When it does go wrong, there appears to be no time-out when it finally will disconnect the line. It does appear to be dependent on the type of system that is called. My earliest tests were using a cell phone as the redirected target. If the call was answered, and then the far end (cell phone) hangs up or “ends call” on the cell phone – all is well – the Norstar resets and is ready for more calls. This is true with almost any call to any type of phone where the far-end recipient answers live and then hangs up first. If the cell phone is off or out of range and you reach the cell phone company system message “sorry your party is not available…” the cell phone company apparently never hangs up the call. It also seems to happen when reaching some other automated systems – but not all. So the line from the line poolA is in use indefinitely until I physically break the cross-connect between the ATA and the trunk port.

To attempt a summary: My initial conclusion is that the problem is due to the far end not hanging up or not registering as already having been hung up. If the call is successful and reaches a live far-end party and the far-end hangs up the call, the call is ended properly and the system is ready for another call. If the near-end hangs up first, but the far end leaves the call off-hook, the system will free up the inbound line but will not free the outbound redirected line.

I have verified that all phone lines involved are set for Supervision. I can break the connection by disconnecting the cross-connect between ATA and trunk port. I have tried adjusting the settings under Capabilities - ATA Settings, but this doesn't seem to effect anything.

Thanks again,
Ted

 
Sorry Ted ...
Glad to help!

Hmmmm .... no DTMF. Are both trunks optioned for Tone dialing?

Damn! Disconnect Supervision!

This is an inherent problem with the ATA's. They just don't provide Disconnect Supervision. I assumed we were sending the call to a "live" person on the redirect. But, the symptom doesn't really point to the ATA if its the Telco trunk being locked up and not our little "virtual" trunk. The flip side is the fix does point to the ATA. The only ATA, compatible with Norstar, that I know provides DS is the Linx ATA. You may be able to Google it and find one.

Any chance of using the VM option I detailed earlier?
That may provide the DTMF signaling, and might have an affect on the DS.

Could be worth a test run.

SD

 
You need a service card installed in the system to provide DTMF tones

Jeff
 
Jeff ....
Services card provides DTMF receivers for E&M T1.
The tone generator is in the core.

SD
 
Most efficient way I've done this is either using PRI or analog 3-way-calling-transfer through the telco. Otherwise known ans hookflash transfer in a PBX. Where I've got analog loops and 3way calling activated my external transfer dial string looks like this:

(route access number) F71 xxx xxxxx F71

THis will connect both parties directly eliminating KSU invlovement in the disconnect tear down. Volume issues are also taken care of. Where ld is concerned the ICS will allow 23 digits in the external dial string so a 1800 number is possible. On one site a CCR destination point in a CC is set to hookflash transfer to a Vonage account with unlimited ld and no disconnect problems as cbc is set up just fine at the near end.

Hope this helps.

PhM



VVV
 
We Have Set This Up On A BCM 400
Call Come Into The AA
You Dial 4 To Go To The Remote Office

The System Will Dial 8123 ( Use Line Pool C )
Set Line Pool C If You Get 8123 Strip 4 Replace With Number Dial Out Number.

ISDN Has The Most Control Of This Divert.

A-log Line Can Lockup

Chris

 
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