Welcome to the Cisco Networking Professionals Ask the Expert conversation. This is an opportunity to discuss with Cisco expert Tejal Patel about the configuration and troubleshooting of all versions of Cisco CallManager Express and its features. Tejal is a customer support engineer at the Technical Assistance Center (TAC) at Cisco Systems, Inc. He joined Cisco in July 1999. His current responsibilities include troubleshooting complex issues, training, and authoring documentation. His areas of expertise are Telco Signaling, Configuration and Troubleshooting of Access Servers, AAA etc. Tejal is a CCIE # 6619 for ISP Dial. He continually shares his expertise by speaking at the Access Design Clinic at Networkers to discuss and resolve design related technical issues.
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Do you know when the next release of CME will be out and what key new features it will include?
The next release of CME, which will be 4.1, is currently scheduled for mid March 2007. We are still in process of compiling the list of features that needs to be included for that. Final new features list will be posted via release notes for CME 4.1.
CME 4.2 which has a target date of June 2007 (as of now) will have a support for IPCC Express. The ios will be 1st release of 12.5T.
Many customers asks for extension mobility service with ccme, is it possible? Or there is a feature that can play a similar task??
I would like to see the following features in the development plan:-
1. support for SRST in CCME to support multi-site SME's with very small branch offices.
2. CiscoTSP that uses the Microsoft username/password as the login parameters for computers that are used by more than one user.
3. an application (similar to CRM Connector) for exchange/outlook contacts.
4. a fully SIP compliant CCME
I know this encroaches on CallManager territory, but CCME will need better application integration if it is to compete with the likes of IP Office.
Can you control phones QoS settings from CME, so that SCCP and RTP DSCP values would be configured centrally (same as it works in call manager)?
There is no way from CME to control the SCCP and RTP DSCP values for the ip phones just like CallManager. The default value is 5 set via config file for the phones anyway.
But you can configure the switch or a router using "policy-map" to change the setting for the packets in and out of the ip phones.
I haven?t done much with CME, customer is asking how they can setup CME so when a 7960 user dials 4XXX it translates to 9,624-4XXX, sending it out across PSTN to another office.
I would do this with a translation pattern in CallManager.
Any running-config code sample appreciated -jason
My users are experiencing dropped calls when trying to place calls to the PSTN. I have FXOs using loopStart signaling. Is there anything that I should be looking for?
First, you need to confirm the signalling type with the telco provider. After that we need to troubleshoot this issue to find the direction (pstn or ip phone side) from which the call is getting disconnected. Here is the url which will help you find that.
You need to turn on the debugs for following
debug vpm signal
debug voip ccapi inout
You can manipulate/translate the calling number and the called number using "voice translation rules" on CME. Here is the url which talks about how it works and how to configure etc. on CME or any voice gateway.
You can acheive this using voice translation rules from your gateway. An example:
voice translation-rule 1
rule 1 /\(4...\)/ /9624\1/
voice translation-profile PREFIX
translate called 1
dial-peer voice 100 pots
translation-profile outgoing PREFIX
you're basically taking the 4-digit DID and prefixing the digits to make it a FULL E164 call.
Hope this helps....
Do you know something about the SCCP client for Nokia Serie E?
I'd like to test it with Cisco unified CME 4.0.
Cisco has licensed Station Call Control Protocol (SCCP) to Nokia. Nokia is developing an SCCP client on its dual-mode Nokia Series 60 devices, which can then be configured and operate as a Cisco Unified IP Phone in a Cisco Unified CallManager network solution.
The SCCP client application builds upon the Cisco Compatibility Extension work that Nokia has done already with its 9500 product. In addition to enterprise WLAN data access, Nokia will provide enterprise WLAN voice access, giving workers mobile access to voice and data features and applications in the enterprise.
You can find more on this at following location
by searching for "Nokia"
the link that you send me is a "broken link"!
However, you saying me that the SCCP Client will operate with Cisco Unified CallManager; not CCME?
I have two more question:
1) When will be released the Cisco Unified CallConnector for Cisco Call Manager Express? The download link on cisco web site is a broken link.
2) Is the ip telephony softkey customizable?
For example, can i configure the call forward noan, call forward all ecc. ecc. via softkey button?
Thanks for all.
Is there any plan to add a service to lookup the calling number(ANI) in the local directories or speed dials and display the associated name for external (PSTN) calls? I would anticipate it would work much the same as the called-name display feature. I have a tcl script doing it now but it uses a separate number/name list.
You can configure the local directory in CME (including the external name and numbers) and once the call from them is received, the names will be displayed accordingly. Here is the url for more on that
Thanks for the information. This appears to only work when the carrier does not send any display-ie. Is it possible to over-ride this? My carrier sends "MOBILE" as the display-ie in the Q.931 message for mobile numbers which seems to stop the lookup of the local directory.
I have a question about Dial-peer management.
Currently we have 60 + sites and dial-peer management has become a huge challenge. Making sure that every router has the most up to date and accurate dial-peer list is almost impossible.
In this particular network we have about half CME running 4.0 the rest are older 1750's with just 2 analog FXS ports (not running CME).
What is the ideal way of setting up a centralized system of managing a dial peer table? I have read that there is a ciscoworks module that manages this and pushes updates to the router.
I have also read that a gatekeeper setup would work as well.
If a gatekeeper will work for both my CME and non CME sites. What is needed to setup a Gatekeeper? Can I purchase the Gatekeeper IOS and load it onto a 2600 series router
Any other suggestions would be greatly appreciated.!
Ciscoworks will manage the routers individually. You can configure it in such a way that it will push the dail-peer part of the config in to all the voice routers that you need.
You can use gatekeeper to simplify your call routing decisions. With that Gateways are responsible for edge routing decisions between the Public Switched Telephone Network (PSTN) and the H.323 network. Cisco gatekeepers handle the core call routing among devices in the H.323 network and provide centralized dial plan administration. Visit
External directories ?
Hello Tejal from a former cisco TME.
I'm about to go with my first CME project and my question is about external directories, are these supported in CME 4.0 ?
You can use the LDAP directory located on the external server. You need to configure the "url command " under telephony-service pointing to the server with directory listings. Visit following lcoation for more in that
I understand than one would need to use Unified Call Manager for LDAP service, or a third-party application, what else could serve as LDAP directory server ?
The CCME router can send the call details to the CDR viewed in the Callmanager? Example, from an IP phone that dials to the PSTN or other sites through the CCME. It's possible?
could you please describe in short how to do this? What commands we use to enable it? Where are CDRs stored? What is the format of CDRs? Or just paste a URL.
Also, it will help if you can specify what is really CDR server? Is it something like FTP, SYSLOG, TFTP, i.e. service with well-known port number(s)?
Thanks in advance,
1. Can CME support the appending of digits on outbound dial peers (specifically append a # digit to signal end of string to a Sat Phone provider)?
2. Is there a method to support a broadcast hunt-group setup? I need to ring multiple stations simultaneously rather than through a longest-idle, or sequential logic. Any chance that analog station ports can be included in said groups?
1)You can use the "voice translation rules" to manipulate numbers. You will find more on that at following location
2)You can configure ip phones with the same number so that oll of them can ring for the broadcast number. Here is the url for more on that
You can configure Analog FXS ports as SCCP and register them with CME. Create dial-peer with the same hunt-group number so that it will ring the ip phones and also the analog ports with the same hunt-group number. Here is the link for more on that. Check for supported hardware and platforms.