DTMF Transfer

[rsenio]

I've got an extension that I want users to call, they would presented with a voice that would say "please enter the extension that you would like to call" then it would take their extension that they entered and transfer them to it. We are using this for people who have no direct line. I'm unsure of what action I need to use before the assisted transfer action. Help!??!

 

[kaphein]

- Create a new menu action, put in the wav that you want ("please dial the extension now")
- Uncheck all the predefined digits in the Touch Tone tab, add only one entry with the same number of "?" as your extensions (ex: 212 = ???)
- Create a new transfer action
- In this new transfer action the Destination will be "$KEY"
- link the entry ("???") in your menu with the transfer action.

As an option you can put in a timeout in your menu redirecting callers elsewhere if they get caught there by mistake.

 

[rsenio]

That seemed to do the trick. However I have both 3 and 4 digit extensions. I simply added the ???? to the first menu but that didn't work.

 

[CarGoSki]

Use two separate entrys in the menu.

????
???

 

[rsenio]

Thats what my menu is now. All I can call is 3 digit extensions. When I try a 4, it just routes back the menu

 

[dizzyphil]

what is the range of the 3 digit and the range of the 4 digit?

dizzy

 

[rsenio]

We have 3 digit extentions in the 3xx and 4xx range. Our 4 digit extentions start at 2xxx. Below is the layouts that I have tried for the menu:
Timeout
???
2???

and I also tried
Timeout
???
????

 

[dizzyphil]

then make your menu tree look like this

Timeout
3??
2???

dizzy

 

[dizzyphil]

also I forgot

timeout
3??
4??
2???

dizzy

 

[rsenio]

That lets me transfer, but for some reason the 4 digit ones are silent for a 20 seconds or so(we have no on hold music) and it goes to voice mail. Thats what the fall back is on my menu, but I know the extensions aren't busy yet they aren't ringing.

 

[dizzyphil]

You can have the same transfer module for the 2?? and 3?? but use a seperate one for the 4???

dizzy

 

[rsenio]

Whats weird is that it was still doing the same thing. As as test I changed the assisted transfer to a straight transfer and all of them work, except for the 28XX series. Those are SIP devices and it rings once and disconnects on the SIP end. I'm sure thats a whole other story in itself. But it is weird that the assisted transfer is what was causing the problem.

 

[tech2233]

Just curious. What SIP devices do you have connected to the IP Office, and how did you connect them ?

 

[rsenio]

We have SIP softphones and a couple of wifi SIP blackberry's. They are connected through an Asterisk (linux)gateway

 

[tech2233]

And does the Asterisk work well ? Was it costly to set up ?

 

[rsenio]

Yes, it doesn't work too bad. If I was more linux savy I could have set it up myself perhaps. We had a consultant come in and configure the link between the Avaya system and the Asterisk box. Couple grand if I remember correctly. That also included him coming on site, travel time etc. The blackberrys work well because when the user enters the office it registers itself with Asterisk and that extension comes live. And voice mail is re-directed to thier email.

 

[tech2233]

What is the interface between the IP Office and the Asterisk ? Is it through analog station ports off the IP Office into the Asterisk ?

 

[rsenio]

IP office uses the h.323 protocol, and doesn't know SIP. So the asterisk box is just a PC running linux and asterisk plugged into the network. It acts as a gateway for the SIP->H.323 and visa versa. THere is a route on our IP office that say anytime someone dials 28XX it goes to the asterisk IP