Voicemail when VPN goes down

[wturner80]

Hello,

I have two locations. I have centralized voicemail at our main location and the two IPO's are connected via site to site VPN. When the VPN goes down I would like all calls to those extensions to go into voicemail. Right now a caller just gets a busy signal. Is this possible?

Thanks

Will

 

[Duffman88]

I have a site i want to try to configure a similar setup.

I'm going to try to set up a seperate incoming call route for the block of DIDs i have at the far location with a backup call route to voicemail. I dont know that i will be able to integrate directly to their voicemail box but at least to the vm so they can dial by name to leave a message.

Chris
ACA- Implement IP Office

 

[Duffman88]

If you are running DIDs with 4 digits incoming try setting up one the following or you can wait till i set it up and i will let you know if it works or not .

Theory 1:

Shortcode *98xxxx,VoicemailCollect,#N

Set the backup destination for your DID incoming call route to *98#

In theory this will send the calls to the *98xxxx shortcode then take the 4 digits dialed and send the call to leave a message for the user. I use something similar for a Direct to VM.
____________________________________________________-

Theory 2

Another thing which may work, but i havent tested yet is to set your destination route to VM:# . Same concept but done straight through the incoming call route.

Chris
ACA- Implement IP Office

 

[Duffman88]

Forgot to mention that for the # to work in the destination route the Incoming Number would have to be set to XXXX and your extensions would have to be 4 digit extensions.

Chris
ACA- Implement IP Office

 

[Westi]

I have no SCN in my lab but maybe if you send all incoming call for the remote site to the VM via a phantom user, then make a User in the voicemail for that.
Create a startpoint and then just transfer it to the correct extension, if the Vm sees the extension as invalid it will send it to the mailbox of that user.
It works locally but I am not sure about the SCN, I borrow another IPO and try it out later.

Joe W.

FHandw., ACA, ACS

If you can't be good, be good at it!

 

[intrigrant]

Maybe you can use ARS for this.
If the link goes down then IPO will try to find a matching shortcode in ARS.
So if the internal users on any location starts with 1 and all numbers are four digits long then create a shortcode in ARS like this:

SC = 1XXX
TN = #1N
Action = VoiceMailCollect
LineID = 0

All calls will go to the users VM Box. If you just want a message like "this extension is temporarely out of order" then send the caller to a module :
SC = 1xxx
TN = "SorryNoLine"
Action = VoiceMailCollect
LineID = 0

 

[Duffman88]

Well the good news is I was able to send a call directly to VM using and incoming call route and shortcode.

Theory 1 worked fine and routed the call to VM. It gave an error when saving to IPO but I suspect thats because the '#' for incoming call routes was designed to be left as the only argument in the destination field. However it was working fine.
--------------------------------------------------------
Theory 1:

Shortcode *98xxxx,VoicemailCollect,#N

Set the backup destination for your DID incoming call route to *98#

In theory this will send the calls to the *98xxxx shortcode then take the 4 digits dialed and send the call to leave a message for the user. I use something similar for a Direct to VM.
---------------------------------------------------------

The only problem I had with it was when i dropped the Point to Point T1 between my campuses (only for about 10 minutes or so) the IPO still knew the extension, used the default route to and tried tried to send it over the SCN. I just ended up w/ dead air on the line. I think this is just because the SCN didnt fully time out. Anyone know how long it takes for the SCN to time out and return a busy/reorder tone? Or will it remember the users/hunt groups across the SCN and continue to try to send calls to them?



Chris
ACA- Implement IP Office

 

[intrigrant]

This timer depends on the version you are running and/or which parameters are set.
Version 3.0 and older do not have any kind of re-route option so the call will try the SCN route for a minute of five and then the call is dropped.
Higher version all have some kind of mechanism to re-route the call and there can be a timer set-up for that.
Search the manuals for LCR/ARS options, usually these options are programmed on the NoUser.