The intent of this document is to cover topics that Cisco customers frequently ask TAC. It is expected to be a living document, and ideas for FAQ entries are encouraged. Answers may include links to bugs, enhancement requests, documentation, tools, or even 3rd party products.
How do I get new items added to this FAQ?
Feel free to suggest new entries via the Discussion thread at the bottom of the document. As the list grows I fully expect a reorganization will be necessary to group entries into separate categories.
What version of CUCM should I be running?
TAC does not recommend any particular version of CUCM. We do recommend customers be on a version that is still actively being maintained with new bug fixes. This makes it much simpler to get the fix for any new bug you may encounter via TAC SR. As a general rule we also recommend customers upgrade to the latest SU available for the version you are on or are upgrading to. This means if you are planning an upgrade to 8.6 you should be going to 8.6(2)SU1 instead of 8.6(1) (no longer being maintained) or 8.6(2) (SU1 is available).
The following versions of CUCM are still getting patched with new bug fixes via engineering specials (ES). If the end of an ES train has been announced for a given version it will be noted (MM/DD/YYYY).
- 8.0(3) 10/23/2012
- 7.1(5) 06/22/2013
What SFTP servers are supported for DRS Backup and Restore?
The DRS Admin Guide for 8.0 says...
To back up data to a remote device on the network, you must have an SFTP server that is configured. Cisco allows you to use any SFTP server product but recommends SFTP products that have been certified with Cisco through the Cisco Technology Developer Partner program (CTDP). CTDP partners, such as GlobalSCAPE, certify their products with specified version of Cisco Unified Communications Manager. For information on which vendors have certified their products with your version of Cisco Unified Communications Manager, refer to the following URL:
The link that follows is to the Cisco Developer Network.
The same doc references these three SFTP servers that Cisco uses for internal testing.
The important thing to note is that whatever SFTP server you use it is supported by the company or person that wrote it. This is the same for a product certified via the CTDP as for any random SFTP server you download off the internet.
How do I disable music on hold (MOH)/play silence for MOH?
CUCM does not have a native option to disable MOH or select tone on hold as an option. CSCdw68639 is an enhancement defect filed to track a related feature and if you feel strongly that this feature is required for your deploment please contact your account team who can put you in touch with the appropriate Product Manager.
That said there are several options available to get the desired behaivor.
- Upload a 60 second recording of silence to use an MOH audio source.
- This is by far the simplest way to have callers hear no MOH. The downsides are that it still requires an MOH resource, the file has to be uploaded to all MOH servers, and if you use a very short duration audio file you can cause CPU spikes on the MOH server.
- Use multicast MOH but prevent the multicast stream from reaching the endpoints.
- This method requires configuring an MOH audio source, MOH server, and media resource group (MRG) for multicast MOH. This will save MOH server resources throughout the cluster as one software resource is used for every device on hold. Blocking the multicast traffic can be done by setting a very low TTL for the audio source (Max Hops field on the MOH server configuration page) or via ACL on an appropriate router in the network. The device on hold will be told to listen for the multicast audio stream but because it never receives the RTP packets the held party will only hear silence.
- Block access to an MOH server via media resource group list and set the parameters to disable tone on hold.
- This final option is the one that comes closest to selecting silence out of the box. It does require different configuration based on device type.
- MGCP gateways must be configured with the command mgcp timer toh-time 65500.
- SCCP phones and gateways are controlled by the Cisco CallManager Service Parameter Tone on Hold Timer. A value of 0 means a single tone will be played. A value of 20000 means no tones will be played.
- 6608 voice gateways are controlled by the Hold Tone Silence Duration value in the gateway configuration page. This timer determines the length of time between beeps and is in ms (1000 = 1s). Values can be between 0 and 65535 where 0 indicates the default 10s timer will be used.
How do I block caller id for outbound calls?
Many PSTN providers offer a service to block outbound caller-id. This is done by dialing a service code before entering the digits of the telephone number you wish to call.
Some sample service codes for blocking caller-id.
- *67 (US)
- 141 (UK)
If your PSTN voice gateway uses an analog port for outbound calls then blocking your outbound caller-id is as easy as sending the service code your provider supports in the outbound dialed digits. You can use a route pattern in CUCM to allow users to dial the service code directly or even set up a called party translation to substitute the appropraite service code for whatever prefix you choose.
9.[2-9]XXXXXXXXX (discard predot) --> sends 10 digits to the PSTN and does not block caller-id
8.[2-9]XXXXXXXXX (discard predot, prefix *67) --> sends *67 followed by 10 digits to the PSTN, does block caller-id
If you are using a digital or IP trunk for PSTN connectivity then we have the ability to restrict caller-id by setting the presentation level to restricted while sending the full calling party number so that emergency services, etc can see it. Sending a presentation level of restricted will cause the far-end to block the display of caller-id, showing "restricted", "private", or "withheld" in its place.
- For SIP trunks you can control presentation of number and name for all calls on the trunk by setting the Calling Line ID Presentation and Calling Name Presentation fields.
- For H.323 gateways and trunks you can control presentation of all calls on the trunk using Calling Party Presentation field.
To restrict or mask caller-id on a per-call basis you can use separate route patterns as described above. Instead of prefixing a service code to the called party number you can set the presentation level directly on the route pattern.
- This document covers masking and blocking caller-id using presentation and external phone number masks.
How to get a Report of incomming and Outgoing calls in CUCM?
The incoming calls refers to inbound calls that originate outside the Cisco Unified Communications Manager network, enter through a gateway, and go into the Cisco Unified Communications Manager network. The Outgoing calls are that originate on one Cisco Unified Communications Manager network, go out through a trunk, and terminate on a different Cisco Unified Communications Manager network.
To generate the report refer the steps mentioned in following URL:
How do I change the backgrounds on all my phones at once?
Not easily. The creation of customized background images has been documented extensively but still requires the user to go and select the background from a list of choices. The Phone Designer widget makes updating a single phone trivial but doesn't support mass updates to multiple phones. While the API exists to programatically tell phones to use a given background image Cisco has not built this capability into CUCM or any other application. There are 3rd parties that have stepped in to fill this void. While we try to remain 3rd party agnostic here this type of search should help nudge you in the right direction.
This document is a very good resource for creating custom backgrounds and covers most device types.
Is it possible to register SPA 500 AND SPA 300 SERIES IP PHONES with CUCM ?
Natively SPA500 and 300 are not supported in CUCM. These phones supports standard SIP and SPCP (Tokenized Skinny), not SCCP.
These Phones are designed/tested with UC500 only. SPA5xx will interoperate with CUCM in SIP mode for basic features (i.e. make/receive calls, call forward, transfer and conferencing.) Other features may not work properly.
Should I use IP address or hostname in CCMAdmin System->Server?
The System->Server field in the CCMAdmin web page is a very important setting. It can have one of three values:
- IP Address
If your phones or MGCP gateways do not have access to a reliable DNS server then you must use IP Addresses in this field.
If you have a reliable DNS solution in your network or rely on NAT between the CUCM server(s) and the IP phones then may safely use hostnames in this field. If using NAT then hostnames are required for accurate DNS resolution to the external IP Address.
- Fully Qualified Domain Name (FQDN)
FQDNs are rarely used in this field but can be used in combination with DNS and endpoints that have a different DNS suffix than the CUCM servers.
Because endpoints rely on this setting for registration and IP phone services it is very important that you choose the appropriate value for your network.
The last way this setting is used is perhaps the most critical. This value is stored in the CUCM server's database and when this value does not match that of the acual CUCM server then the database will not start. This type of mismatch is most common when changing a server's IP address or hostname. This is why there was a warning prompting you to make sure this field is updated prior to changing hostname or IP address. The CLI commands to update IP address and hostname trigger an automatic reboot and if you have not updated System->Server first then after the reboot no services will be able to start due to the non-functioning database. To recover from this situation you must change the IP address or hostname back to the original value, reboot, then start the process over (this time paying attention to the warnings).
Why do calls to my mobile phone via SNR show my company's main number instead of the real caller?
Single Number Reach, Cisco Unified Mobility Mobile Connect, or Mobility, whatever your like to call it this feature is a very useful one where calls to your IP phone can ring simultaneously on your mobile phone. One problem that customers repeatedly see when deploying this feature is that the call to the mobile phone always shows the customer's main number rather than the real calling party number. The call ringing on the IP phone will show the correct calling party number.
This is caused by some service providers that do not allow customers to send a calling party number that is not allocated to them. It is essentially a caller-id spoofing prevention mechanism. To help combat this restriction CUCM 6.1(4) added a feature to populate the RDNIS or SIP diversion header with information about the IP phone that triggered the outbound call. By populating this field service providers can correctly bill for the outbound call and allow the real calling party number to be sent to the mobile phone.
How to bring the users from two different Active Directory servers if they do not belong to the same Domain. Is it possible with CUCM 7.x?
You would need to use Microsoft Active Directory Lightweight Directory Service (AD LDS), formerly known as Active Directory Application Mode (ADAM), This feature to retrive users from different domains was added in CUCM 8.x and later. In CUCM 7.x, the user is explicitly defined which can only point to one domain such as cn=administrator,ou=users,dc=cisco,dc=com, so it can't bind to more than one domain. For further details referhttps://supportforums.cisco.com/docs/DOC-16356#Requirements
Can TAC help me design my dialplan?
TAC is first a break/fix organization which priortizes getting production networks back online. Requesting design assistance from TAC takes time away from customers with true network outages. How would you feel about waiting in line when your network is down?.
While TAC employs some of best and brightest individuals in the industry our training is focused on troubleshooting and not design. If you have questions about how a feature works or a concern that a product or feature is not working the way you expected it to then TAC is here to help. If you are looking for someone to design a network or dialplan for you then there are other resources much better equipped to help you out. These resources can be Cisco Advanced Services, your Cisco account team, or a Cisco partner.
What version of CUCM is supported on my server?
I'm running version x. What do I need to do to get to version y and are there any "gotchas"?
There are several links critical to planning any upgrade.
Can I run CUCM in VMware and will TAC support it?
As of CUCM 8.0 Cisco is supporting some UC products to run in VMware on Cisco UCS hardware. This has been covered extensively in both CSC documents and cisco.com documentation.
Additionally, CUCM has allowed vmware installs to proceed since 5.1(3). These installs include demo licenses sufficient for training and lab testing. Demo/Eval licenses will be lost if a VMWare CUCM is upgraded but they can be re-downloaded from http://www.cisco.com/go/license. TAC does not support any CUCM installed in VMWare that does not meet the requirements outlined in the links above.
Why am I getting error "The file name contains invalid characters. The valid characters for file name are alphabets, numbers, dash, underscore, dot. The filename should not begin with a dot"
While attempting to upload a file for BAT with Safari on Mac an error message is generated: "The file name contains invalid characters. The valid characters for file name are alphabets, numbers, dash, underscore, dot. The filename should not begin with a dot." The workaround is to use firefox.
You are having your CUCM cluster connected with an H323 Voice Gateway. You need to restrict certain calling numbers when they call to PSTN. your provider can restrict only full pool of your DDI. But you would like to have an option to restrict only some from your DDIs. How to accomplish this task?
The common way to do what you want is to setup a separate CSS that contains a PT that houses the RPs used for routing calls to the PSTN. On these RPs you set Calling Party Transformation to
Calling Line ID Presentation: Restricted
Calling Name Presentation: Restricted
That should restrict presentation of caller ID to the PSTN. It will still show the calling number in the deb isdn q931, but this flag
i = 0x00A3 on the calling party can hide the number. Here in this example the last digit is 3 which can vary.
Is there a time limit to use demo licenses with CUCM?
Cisco Unified Communications Manager contains a starter license(demo license) that you can use to begin new installations of Cisco Unified Communications Manager before you install the production license. Starter licenses, which are available in limited quantities, have no expiration date. You can use starter licenses only for fresh installations; you cannot use them for upgrades or migrations from previous releases. Starter licenses support only one Cisco Unified Communications Manager node and up to 50 device license units.
The system overwrites the starter licence when you obtain and upload your production licenses. Refer https://supportforums.cisco.com/thread/2183206 for more information.