Welcome to the Cisco Networking Professionals Ask the Expert conversation. This is an opportunity to discuss with Cisco expert Markus Schneider the Cisco Unity Express (CUE), its configuration, troubleshooting as well as the customized Auto Attendant scripting. Markus Schneider is a voice network engineer at Cisco Systems, Inc. He has supported Cisco voice products since 1998 and has been a technical reviewer for the troubleshooting Cisco IP Telephony, Cisco Unity deployment and solutions guide, and Cisco IP Communications Express books. He works closely with various Cisco engineering teams and is actively involved in field trials of new products. Markus has also worked extensively with configuring, troubleshooting, and developing custom auto attendant scripts for Cisco Unity Express (CUE).
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im running cme 4.0 and cue 2.3. Is it possible for users to retrieve internal vm from the pstn calling in without the use of a DID on the vm pilot ?
if there's any way to get to the user's voicemail greeting, you should be able to press * to get to the sign-in. There's no menu choice directly in the canned auto-attendant that sends the call to voicemail sign-in.
To expand on that, with a custom auto attendant, you could configure a menu choice which would the use the 'Call Subflow' step to invoke voicebrowser.aef. That will the voicemail system and ask for your ID and password.
Yes, the only difference between those two modules is the memory, which can be ordered. You do need to make sure you're running a 2.3 version of CUE for higher mailbox licenses (which, of course, is extra, too).
Hi, I have a simple question about SRST.
In Cisco SRST v3.3 admin guide, For SRST, we need configure a DHCP pool with Command line "option 150 ip 10.0.22.1" IP address is the callmanager ip address. I thought in SRST mode, the IP phone can not reach the callmanager. So what's this command line used for?
The DHCP scope, of course, is to hand out IP addresses--even when CallManager is still reachable. So you're really telling the phone how to get to the TFTP server, where it will download information about where the SRST router is (as well as other CallManager servers).
Put another way, if you were not using SRST but were still using the router as a DHCP server, you would still need that option for the phones to find the CallManager.
If I have a centrolised DHCP server in head office CallManager site that tell the TFTP IP address for the remote site IP phone, do I have to put the "option 150" in remote site SRST router DHCP pool config?
If you have centralized DHCP that is providing IP addresses for your remote phones, then you don't need (and shouldn't have) DHCP in the SRST router at all. Otherwise there would be two DHCP servers on the same subnet without any knowledge of each other, which isn't a good idea (my guess is that the router would probably respond more quickly because there's less delay, but why take a chance).
I know that you can include a voice mail message in a notification. Is there any way to upload a message that has all ready been made from Unity Express into a wav file or any sound file. I am using CME 4.0 and Unity Express 2.3.
unfortunately there's no way to upload a sound file into voicemail at all. Of course, with auto attendant prompts (and even recorded names for networked users), it's possible, just not with messages. I don't believe that's going to happen in the real near term either (definitely not for the CUE2.3.3 release).
I'm using autoattendant but i'don't know how to associate the "0" (that voice prompts) to the DN of the operator.
My operator has 201 but when voice prompts
and i digit 0 the call is not forwarded to 201!
Thanks for your question. Are you using the auto attendant that ships with CUE? If so, the easiest way is to change the operator extension (the number that CUE calls when 0 is pressed from the auto attendant) is to log into the CUE web page as an administrator, select Voicemail > Call Handling. Then in the Voice Mail Operator Number put in 201 and click Apply. Then it will send the call to the 201 extension.
If this is a customized auto attendant script, then the process may be more complex, as the number should either be set in the script, or--more commonly--exposed such that it also can be changed from the CUE web page. In that case, it might be best if you could send a copy of the script.
Yes, it is. Keep in mind that you have to exit out of configuration mode ("end") entirely for that to take effect.
I've problems with my IP phone (7940 Series). When I go into voice message it told us that "You have no old
messages" but the voice mail indicator still appears (red light) and message indicator still blink. Please show me how to solve this matter.
that can have a lot of causes. What phone system are you using? CallManager or CallManager Express? You may want to take a look at the following document:
What I would do first is log into CUE via the CLI and issue the command:
mwi refresh telephonenumber
That will have the system send the MWI again (which it should every time you check messages anyway). If it fixes it, I would try leaving a message--make sure the light comes on--and then retrieve the message--make sure it turns back off. If so, then it was probably some error that happened right when the light was supposed to be turned off.
If that doesn't work (more likely), then there's something else, which would make it important to look at the tech note I mentioned above.
I would need to know if it is using CCM or CCME first and it might be helpful to see the CUE configuration.
I have a DSP problem with my voice router. DSP calculator result is (optimized) PVDM2-32 & codec complexity flex. but an error is comming that "no DSP found locally or globally" all voice port is not working.
what canI do now ?
I think this question would be better suited in the IP telephony forum here:
This particular discussion relates to the Cisco Unity Express product.
I don't believe Russian has been officially committed yet to the product at this time. I would contact your cisco account team so they can drive the business case with the product team. At this time, it definitely won't be in the 2.3.3 CUE release, but it's possible they might be able to put it into a subsequent release or so.
Good question. Currently in the CUE auto attendant, there is no way to look up a user by first name first. The only method is last name followed by first name.
You can swap the first name / last name entries for your CUE subscribers, however, the voicemail system does not allow the addressing prompt to be changed, so in voicemail it will still say to address the last name followed by first name, even though the lookup will now be using first name first.
In short, there's no way to make the auto attendant and voicemail to consistently use first-name-first.
There is an outstanding feature enhancement (CSCsc96431) which is supposed to fix this, however I have not heard in what timeframe that will occur. If needed, you should open a TAC case to reference that enhancement ID so the product team can properly prioritize the feature.
I'm running CUE 2.3 and when a caller leaves a VM it lets them either hang up or hit # for more options... nothing new there... however, if they hit # it let's them specify urgent or normal priority then it automatically sends them to the operator. Why are they transfered to the operator and is there a way to just have the call disconnected?
Beginning in CUE 2.3 you should be able to configure a null VM operator number and thus stop this behavior (you can still configure individual zero-out numbers for during the greeting).