Unity Question

[admin_2]

Quick question: Just installed a site, integrated with Call Manager 3.05a. I have the latest tsp from Active Voice and I am doing supervised transfers from the auto attendant.<br><br>When outside callers come in and access the AA they succesfully get transfered to users, however they are unable to leave messages. They cannot break through the greeting, hitting # just causes the message to be repeated. A sample is to hear the greeting, hear the beep, silence for a second then repeated greeting, etc. <br><br>When users call each other internally there is no problem.<br><br>Any clues? I think I may have a call handler messed up. Is this default behaviour for extended absence or something?<br><br>Thanks,<br><br>Jim<br><br>

[#2]

Well… if internal callers are behaving properly I would tend to suspect your call handlers are setup properly.

If you’re hearing the beep, then it’s going into record mode… you may have your after message action set to go back to the same call handler I suppose, though I doubt it since your internal callers aren’t experiencing the same thing (I’m assuming) and there’s only one “after message action” setting for each handler. The # key is normally mapped to “skip greeting”, you can check to see if that’s the case… it could be mapped to cycle back around, although again that’s not the default configuration.

This is one of those things I’d need to look at how you have it setup to be more helpful…


Jeff Lindborg
Unity Product Architect
Active Voice
jlindborg@activevoice.com
http://members.home.net/jlindborg

[#3]

I'm having a similar problem... Has there been any resolution to this yet? Please review the follwing forum post below that I made earlier today...

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I'm integrating ActiveVoice Enterprise with the Cisco ICS 7750... I've got everything working very well with the exception of one thing...

When I place an external PSTN call to the ICS7750 system through an FXO port, the call is successfully transferred through CallManager to ActiveVoice where the subscriber's greeting message is heard... During the greeting message the user is prompted to enter an extension or 0 for the operator.

The problem is that any button presses made from the originating external PSTN side phone are not heard or recognized by ActiveVoice during or after the greeting. When placing an internal IP phone call to the same ActiveVoice subscriber, the greeting again is heard and the button presses are successfully recognized by ActiveVoice - everything works as expected...

I have the FXO port set up as a POTS dial-peer in the Cisco MRP manager... I know this is functioning because otherwise I wouldn't be able to hear the ActiveVoice greetings, etc.

It seems that the audio coming from the originating PSTN side phone is not recognized until the call is transferred out of ActiveVoice and back to a Cisco 7960 IP phone extension... In other words, it seems I'm only getting one-way audio during the ActiveVoice connection from a POTS dial-peer...

ANY HELP!?!?!?!

[#4]

Oh yeah, I almost forgot...

Brian Radmer
Integrated Telecommunication Systems
2775 Algonquin Road
Rolling Meadows, IL 60008
847.368.8400
bradmer@itsinfo.com

[#5]

I am having the same issue with one of our call handlers. One is setup to allow direct dial-in to the voice mail system. If users dial internal, the dtmf tones are accepted. From outside, they are not. I have checked the caller input and all seems to be in order. I thought perhaps it was a gateway issue, but dtmf is working in the system from the main greeting etc., it is just not working under this call handler. If you need any other information, please let me know.

[#6]

If you access this call handler from an interanal extension and it works properly, the handler is setup OK and the problem must be on the switch/gateway end of things... call handlers don't (and couldn't even if we wanted to) make a distinction between caller input from internal vs. external sources... DTMF is DTMF to a call handler.

running the integration monitor with the "show digits" option in the view menu enabled should show what DTMF we're getting. I'm betting on an internal call you'll see DTMF getting recorded and for external calls you wont see anything.

Jeff Lindborg
Unity Product Architect
Active Voice
jlindborg@activevoice.com
http://members.home.net/jlindborg

[#7]

Because this is a Call Manager integration, is there another tool to use to see dtmf tones?

[#8]

The integration monitor should actually show digits pressed even with IP integrations. If you select "include digits" in the View menu, it should should you what's being pressed on any given port.

If for whatever reason that doesn't fly for you (do let me know if that's the case) you can also use "StatusMonitor.exe" found in the \CommServer\TechTools directory. It's a little busy since it's showing lots of different things such as conversation state changes and the like, but this will always show you what's happening on a port. When you fire it up, make sure to check the "conversation" checkbox in the "monitor settings when started" box on the right. Fire up all the ports or just the one you're testing with and it should show you what happens when a digit is pressed during the conversation. If you don't see any activity, we aren't getting digits.



Jeff Lindborg
Unity Product Architect
Active Voice
jlindborg@activevoice.com
http://members.home.net/jlindborg

[#9]

After looking into your recommendation, I have noticed a few things. First, if I outside call directly into the opening greeting, DTMF tones work fine. Outside calls direct into a DID to a users voice mail, dtmf are not continued. Any ideas?

[#10]

Chatting with some of our CM savvy folks around here, they suggest that you need DTMF relay capable/enabled on your gateway... Aaron passed along the following info:

Here is everything you wanted to know about dtmf relay and more. This was taken from the IOS documentation online.
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121cgcr/multi_r/mrd_cb.htm#xtocid1872354

You want to set the "dtmf-relay h245-signal". Something like this.
configure terminal
dial-peer voice 103 voip
dtmf-relay h245-signal
end

If the site is not getting dtmf's from the gateway, then they dial peer they using for the call does not have this value set. The engineer probably setup the Unity dial-peer with the proper dtmf-relay value. They probably did not setup the dial-peer for the DID lines properly.


Jeff Lindborg
Unity Product Architect
Active Voice
jlindborg@activevoice.com
http://members.home.net/jlindborg

[mslaga]

Add DTMF-Relay to your VG200 or whatever gateway you are using. This should fix the problem.

Matt Slaga
TimeBridge Technologies
MCNE, MCSE2k-4.0-3.51, CCNA, CIPT2.0