Calls not rerouting to greeting.

[Volguy7]

I have a proprietary system we use internally for speech-driven dial-by-name. it is an analog system with a pilot DN and 4 analog ONS stations in a hunt group. When you speak the name of your party, it does a hook flash and dialas this party, so it's a transfer. The issue is that when this system calls a party that is not available even though it rings the extension the call reroutes to Auto Attendant. If the called party is available, they can answer it live. Now if I dial any station in the system directly from my Multiline IP set, the call reroutes to voicemail, but I hear the individuals voicemail greeting. All of these analog stations are in the same COS as my IP sets. There are no interconnect restrictions. What am I missing and why will calls transferred by my speech system not reroute to the called aprties greeting but instead reroute to my auto attendant? Thanks in advance.

 

[kwbMitel]

It would help if we knew the Mitel System Type.

As the 3300 does not handle analog inband integration very well, I will assume you have a 200.

I would watch the status or monitor the voicemail ports while they are making the transfer. The symptoms would be easily explained if the voicemail is doing a supervised transfer instead of blind transfer and it doesn't know what to do when the call routes back to itself.

If it is doing a blind transfer, I would need more details about the system and integration method before speculating.



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Consistancy... Is only a virtue when you're not a total screwup.

 

[Volguy7]

How could I omit something so simple? Mitel 3300 LX Rls. 7.0. The system that is transferring calls is using a Dialogic D/4 PCI analog card. The Mitel Circuit Descriptor has no changes from default. One further step I took was connecting a test set to the lines conencting to my speech server and dial an extension and it rerouted to voicemail. So the issue is definitely when my speech system does a hook flash and transfers to voicemail. It is so strange because the transfer is successful, it just transfers to my Auto Attendant instead of mailbox greeting. I also noticed that it of course is hitting my voicemail ports.



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[cwc3]

do you need a * prefix + mailbox number to bypass
auto attendant?

 

[kwbMitel]

something is missing. With analog voicemail integration you must tell the voicemail where the call is coming from, ususally with DTMF digits. You have not mentioned how this is done. Simply rerouting to the hunt group master does not make sense. We need details about the integration.

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Consistancy... Is only a virtue when you're not a total screwup.

 

[Volguy7]

I was not aware that an incoming Analog call to a Mitel voicemail ports was different than a digital one. So if I picked up an analog phone in my system and called a co-worker that was not at his desk, I would not reroute straight into his greeting but instead into to Auto Attendant. This is why I am am confused with your integration questions. I'm not integrating anythign with voicemail, I am merely using an analog extension to transfer a call into a voicemail box and it is sinstead rerouting into Auto Attendant.

CW3 - Not with my digital sets, it goes straight to voicemail greeting. I can choose to do that from the Auto Attendant.

Thanks for the help.



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I'm new, but I'm learning.

 

[Lundah]

I agree with kwb here, for some reason the VM isn't getting the integration digits when a call originating from the analog ports is re-routed to voicemail. I have a hunch it might be a COS setting on the analog ports, can you tell us what the COS settings are for the analog ports for the following fields?

COV/ONS/E&M Voice Mail Port
Disable Call Reroute Chaining on Diversion
Follow 2nd Alternate Reroute for Recall to Busy ACD Agent
HCI/CTI/TAPI Call Control Allowed
HCI/CTI/TAPI Monitor Allowed


Also, how are the calls getting to the analog ports on the speech server? Through trunks directly connected to the same controller, or is this over a network? What type of trunks?

 

[kwbMitel]

Ok, Maybe I've been not understanding the environment completely.

My understanding now is you have 3 elements to this problem.

1 - the SX200 Controller
2 - An embedded Mail or Express messenger
3 - a proprietary autoattendant with voice recognition.

If this is not the case, please explain.

Assuming it is, I would insert an analog phone in place of the Auto-Att and manually answer and transfer the call. This will eliminate anything the Auto-Att might be doing to cause the problem. Make sure that the Analog set is set up identically as the Auto-att, COS, COR and Inbound routing. Test with external calls.

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If we don't take care of the customer, Maybe they'll stop bugging us.

 

[Volguy7]

Lundah -

COV/ONS/E&M Voice Mail Port (Yes)
Disable Call Reroute Chaining on Diversion (No)
Follow 2nd Alternate Reroute for Recall to Busy ACD Agent (No)
HCI/CTI/TAPI Call Control Allowed (No)
HCI/CTI/TAPI Monitor Allowed (No)

These are ONS stations on an ASU II. I have a hunt group:

582 with the following stations
3090
3089
3017
3045
The Hunt group is using the same COS of the stations. On most of my tests, I've been dialing the extension of the hunt group as opposed to the DID of the Hunt Group, the results are the same with internal versus external calls.

KWBMitel - I took the Auto Attendant out of the equations. I put an analog set on the first ONS channel. I dialed the extensions 3090, with another phone. I answered, did a hook flash and got dial tone back, then dialed an extension which I knew would not be answered. Once I heard it begin to ring, I hung up the ONS stations so the calling party was conferenced with the called party (the ONS was out of the equation. Instead of hearing the users greeting, I got the auto attendant.
Next I dialed the same extensioon directly with the ONS station and after the set # of rings, heard the user's greeting.
So it seems as if any call transferred to my voicemail ports by an analog station will only get the auto attendant, not the user's greeting.
Thanks for everyone's time and patience.






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I'm new, but I'm learning.

 

[kwbMitel]

Hold On, Don't give up yet.

Based on your test, I believe the call is retaining the ID of the autoatt group as the Originally Dialed Extension and trying to integrate with that number.

As you probably do not have that number as a mailbox the call ends up at autoatt.

This fits the facts as I know them at least.

The next step is to take the VM out of the picture. Set up a test phone (multibutton display) to act as your VM. Route a phone to this "Test VM" the same way you route to normal voicemail.

Perform your tests and observe what is displayed on the screen of the set.

You are looking for "FROM XXXX"

Do both internal and external tests and note if there is any difference in what is displayed.

Interested to find out if they are different.

*******************************************************
If we don't take care of the customer, Maybe they'll stop bugging us.

 

[Volguy7]

Thanks for a great response. I tested internal and external callers with a display phone as my "voicemail". In both cases, it shows the calling party transferring the call as my pilot DN for the hunt group. So call calls coming out of that hunt group show as originating from the hunt group, no matter what the DN is. And apparently calls to VM from the hunt group are not passing on DTMF digits to show the extension being dialed.


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You can't fix stupid, you can only work around it.

 

[Volguy7]

I hate to say it was such a simple thing after all this trouble but it was. The hunt group type was NameTag instead of Voice. Now it works and I know not to use Nametag.


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You can't fix stupid, you can only work around it.

 

[kwbMitel]

Glad to help, learned something in the process as well.



*******************************************************
If we don't take care of the customer, Maybe they'll stop bugging us.

 

[Volguy7]

I learned a lot as well. I'm an application support guy who's ended up building 3 resilient 3300 ICP's with ACD, MLAA and Teleworker and coming soon....SIP Trunking. Not bad, but you and Lundah are by far the most knowledgeable two in a forum on the net. I'm sure I'll post again. Thanks again everyone.


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You can't fix stupid, you can only work around it.