We are replacing an old Siemens system that has an option to leave a message for a user without ringing the set.
I don't want this to be an all the time thing, so I don't want to set the transfer rules to never ring. I would like to create an option off a call handler that gives the caller an option to select a mailbox - by extension or from a Directory Handler - to leave a message for, record the message, then send but never ring the phone.
Is this possible with Connection?
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Thank you for the reply.
I understand you can just send transfered calls directly to greeting, but doesn't that then make any call transfered to that user ALWAYS bypass ringing the set?
I want this conditional - like from the directory handler, giving the caller the option to leave a message for a unity subscriber without hitting their set. Not an all-the-time thing.
Are you asking about the ability to dial the user from another phone and bypass the ringing and go directly to the user's greeting?
If so, the way this is typically done is by building a CTI Route Point with DN *XXXX, where XXXX represents 4 digit dial plan, then having a VM Profile assigned to this CTI RP that masks the number as XXXX (stripping *) and setting call forward all on it to voicemail.
This way if my DN is 5001 and you dial *5001 it will go straight to my voicemail.
You can use anything else you want in place of *, this is just an example.
We've built the *XXXX to VM one as well and that's working.
Here's the flow as it works in the Siemens System.
Outsider Caller > Main Number Handled by an Auto Attendant
Caller is presented with the AA Greeting and given an option (say opt4) to leave a message for a User
Caller presses 4
They are given the option to search the directory for a user to leave a message for
the search for Jane Smith at x 1234
They are asked to leave their message at the tone, then press #
The message is delivered directly to the subscriber without ever rining the set
After # they are asked to search for another user, or to return to the main menu.
They are looking to replicate this feature.
What I provided in the first response is exactly that. The transfer setting controls whether the call will go to the user's mailbox or ring the extension. What you cannot do is have it both ways, i.e. leave a message or ring extension when the call comes via AA. This is all assuming the AA is built in Unity Connection and not another app i.e. UCCX.
Thank you again - Your first answer does indeed answer the question if I was looking for a 100% of the time answer. The customer was looking for a 50/50 answer and does indeed want it both ways.
So, without a UCCX application the answer in this particular case is 'no'
I know this is three years late but:
I've had to setup a call flow where users dial in, hit a call handler, and want to dial-by-extension directly to a voicemail from that call handler. (without breaking the ability to also use regular dial-by-extension).
What I did was created an extension on a cti that is "*xxxx". This CTI will have a Vmail profile that passes the extension (xxxx).
Then, after building the above call flow that they will call into:
Dial Plan > Search Spaces - Create a new search space and assign NO partitions.
On that call handler, under caller input, have the "Prepend Digits to Dialed Extensions" be "*".
Then, under the basic configuration of the call handler, apply your new search space to its "search scope".
Now, go into System Settings > Restriction Tables
Under "Default Transfer" and "Default System Transfer" add a rule to allow "*????"
The caller will dial an extension and it will be prepended with "*"
The reason we apply no partitions is that the prepended entry will now try to route on unity. If it matches 0 users then it will remove the prepend and then try to route/transfer to the user (just the extension) handset as per normal function. But, since we removed all partitions, it will find 0 users and then just transfer the call to *xxxx. Your restriction table entries will allow this through.
Your original SIP leg will be SIP Referred to the new *XXXX that they dialed. This will then trigger the CTI we made at the start, transferring the call to VM with the requested user dialed as the target. be sure that your CTI's extension (*xxxx) is in a partition that is in a css on your inbound trunk and unity trunks "Rerouting Calling Search Space".
Sorry for any typos - typed this up quickly.
I don't follow why you needed to do all of this? What was your call flow? If this was simply to allow users from Cisco phones to reach other Cisco user's direct voicemail then the only thing you need is CTI RP with DN *XXX (VM profile set to XXX), nothing else is needed as by the time the call reaches Unity it will be presented with DN XXX which will match the user's extension.
You'll notice my understanding of what you explained since it is part of that flow but:
They needed to be able to do this from outside of the system, call in from the PSTN, while not breaking normal DBE.
Cody - on the CTI route point, how is that configured? My call flow is incoming 10 digit -> CTI DN which is set to CFWD to unity. Then goes to a call handler. When I enter the extension, unity comes back with "I'm sorry I don't recognize that extension" - this happens for any extension. Remote port monitor is not revealing. Any suggestions?