This blog will cover running CUCM and/or CUCME for a home lab environment. I put together my home lab over seven years ago when Call Manager 3.0 could not be run on a non-Cisco approved platform. The only way to get Call Manager to install was to first install Windows 2000 server, SQL server 2000, and the required Microsoft service packs. I decided to buy my own Compaq DL320 at the time, but I highly recommend using VMWare nowadays. We’ll get back to VMWare…
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Over 7 years ago, I started teaching classes at Global Knowledge. Many of you reading this blog, would like to have your own test environment to play with so you don’t blow up the production network. I invested in a home IP Telephony lab setup so I could keep up with changes and test options not in use at any of my customers. I will cover most of the facets of creating your own Cisco IP Telephony environment in this blog.
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Translation Patterns in CUCM are very similar to Route Patterns, but they do not have a Call Routing destination. Route Patterns point to Route Lists or gateways, while translation patterns are used to manipulate digits. Translation patterns are used to Block calls in the Line/Device Cisco Class of Service approach. Translation patterns are processed as Urgent Priority by default in the CUCM call routing database. The Urgent Priority check box cannot be de-selected. Translation patterns leverage Calling and Called Party Transformations to perform digit manipulation. It is assumed that we are already familiar with these methods of digit manipulation from previous blogs.
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In the last blog, we explored an inter-site call routing scenario where 5-digit dialing was used over the IP WAN between Cisco Unified Communications Manager clusters (trunks) and 11-digit dialing was used to route inter-site calls over the PSTN.
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NYC DID: (212) 551-XXXX
SJC-NY Route Pattern: 55.1XXXX
SJC DID: (408) 622-XXXX
NYC Route Pattern: 62.2XXXX
The San Jose (SJC) cluster Route Pattern of 55.1XXXX is pointed to the New York City Route List (NYC_RL) which has a primary call routing path over the Wide Area Network (WAN_RG) Route Group and a secondary call routing path pointed to the Public Switched Telephone Network (PSTN_RG) Route Group. The Cisco Unified Communications Manager at San Jose only needs to receive the last 5 digits to properly route the call to a Cisco IP Phone registered to the San Jose cluster.
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Continuing our call routing discussion, let’s move on to Route Lists now that we have discussed Route Patterns, Route Groups, Gateways, and Trunks.
Recall that the logical processing order of these configuration elements are as follows:
Route Pattern -> Route List -> Route Group -> Gateway/Trunk
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The last blog introduced the four trunk types used in Cisco Unified Communications Manager (CUCM) to route calls over an IP network. The four types are as follows:
• Non-Gatekeeper Controlled Inter-Cluster Trunk
• Gatekeeper Controlled Inter-Cluster Trunk
• H.225 Trunk
• SIP Trunk
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The NANP dial plan has been discussed in the past four blogs. By default, Cisco Unified Communications Manager (CUCM) does not know how to route any calls to the PSTN based on the NANP or any other International dial plan. All phone numbers that are external to the CUCM cluster must be programmed in CUCM. PSTN phone numbers are configured as Route Patterns in CUCM. These phone calls are normally routed out a gateway for PSTN access.
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In the last blog, we discussed 11-digit phone numbers that carry International tariffs. These route patterns can be removed from the Cisco Unified Communications Manager (CUCM) dial plan by configuring individual route patterns with a Block This Route Pattern condition. Each route pattern to be specified was in the last blog entry.
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The dial plan discussion continues…
• 911 Emergency Call Routing
• 9.911 Emergency Call Routing
• 9.[2-8]11 Three digit service codes
• 9.[2-9]XXXXXX Local 7-digit dialing
• 9.[2-9]XX[2-9]XXXXXX Local area code 10-digit dialing
• 9.1[2-9]XX[2-9]XXXXXX Long Distance 11-digit dialing
• 9.011! International dialing
• 9.011!# International dialing
• 9.0 PSTN operator
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In the last blog, we began analyzing an example PSTN dial-plan. The dial plan has been provided below to make it easier to cross reference the explanation that follows:
• 911 Emergency Call Routing
• 9.911 Emergency Call Routing w/access code
• 9.[2-8]11 Three digit service codes
• 9.[2-9]XXXXXX Local 7-digit dialing
• 9.[2-9]XX[2-9]XXXXXX Local area code 10-digit dialing
• 9.1[2-9]XX[2-9]XXXXXX Long Distance 11-digit dialing
• 9.011! International dialing
• 9.011!# International dialing
• 9.0 PSTN operator Most organizations have, and continue to use, a 9 to access an outside line.
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One of the most important aspects of any voice system is the call routing component. Cisco Unified Communications Manager (CUCM) call routing uses four important components:
• Route Pattern
• Route List
• Route Group
• Gateway/Trunk
A route pattern is a configured numeric pattern range. After the route pattern is matched, the call is routed to the final destination. Route patterns can have various wildcard patterns as follows:
• X Matches one single digit (0-9, *, or #)
• ! Matches one or more digits
• [x-y] Matches one single digit within a range of possible digits
• [^x-y] Exclusion range. Matches any one single digit other than the range of configured digits
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It has been brought to my attention that I have not formally introduced this blog to the readers. I would like to use this blog entry to do so.
This blog is dedicated to conversations related to Unified Communications. The focus will be on Cisco Unified Communications, but we will also discuss integration with other popular Unified Communications platforms like Microsoft Live Communications Server,
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The last blog entry covered DTMF relay. This blog entry will focus on Cisco IP Phone and Cisco SIP Trunk DTMF relay support. Cisco’s first generation of IP phones do not support RFC2833 DTMF-Relay. All digits are sent out of band in either SCCP or SIP signaling packets depending on the phone load. Cisco first generation phones are referred to as Type A phones in the Cisco Unified Communications Manager 6.x SRND (Solution Reference Network Design) guide available at www.cisco.com/go/srnd. Type A phones include the following phones:
• 7902
• 7905
• 7910
• 7912
• 7940
• 7960
Cisco’s second generation phones support RFC2833 DTMF Relay when registered with Cisco Unified Call Manager versions 5.0 and later. These phones include the following:
• 7906
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Dual tone multi-frequency (DTMF) signaling is used on traditional analog plain old telephony service (PoTS) phones to send digits to the switching center for call routing or some other specialized function. DTMF digits are also used to specialized functions like voice mail login, automated attendant (AA), and interactive voice response (IVR) systems. DTMF digits are frequency combinations that enable tone dialing on the PSTN. Pulse dialing was historically used with rotary phones. Voice over IP systems have many methods to leverage DTMF digits depending on the signaling mechanism used.
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This document will cover some of the comparison differences when integrating either MGCP or H.323 gateway protocols. The last blog entry compared the survivability differences of the two gateway protocols.
Analog FXO gateway voice ports using MGCP did not support the Caller-ID service. Caller ID and Automatic Number Identification (ANI) services have been supported over MGCP controlled T1-CAS (Channel Associated Signaling) and T1 PRI (ISDN Primary Rate Interface) voice ports for a long time. Cisco IOS 12.4(15T) code has the caller ID service, but earlier versions of IOS do not. The 12.4(15T) code features are in the XJ code train as well [e.g. 12.4(15T)].
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When considering the use of Media Gateway Control Protocol (MGCP),. H.323, or Session Initiation Protocol (SIP) gateway protocols, feature support should be a consideration. This blog will cover the call preservation feature.
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Skinny Client Control Protocol (SCCP) is a proprietary protocol line side protocol that was originally created by Selsius. Cisco acquired Selsius in 1998 for 145 million in cash and stock. The newly acquired Selsius Call Manager platform used a standard Intel based PC architecture and all connections to time-division multiplexing (TDM) resources were accomplished using PCI expansion cards. The platform had analog cards (trunk or station) and gateway interfaces (T1 or E1) that all used a protocol called the Skinny Gateway Control Protocol (SGCP).
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Media Gateway Control Protocol configuration was configured manually in much older versions of IOS. The manual configuration required extensive configuration that has been automated with IOS 12.2(2). Use the Cisco IOS Feature Navigator at www.cisco.com/go/fn to determine the exact version of IOS and feature set required to support the automatic ccm-manager config download feature.
The automatic MGCP provisioning feature will automate the global MGCP configuration with the use of the following two commands:
ccm-manager config
ccm-manager config server 10.1.1.100
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Various gateway protocols are supported with Cisco Unified Communications Manager (CUCM). The gateway protocols include H.323, MGCP (Media Gateway Control Protocol), SIP (Session Initiation Protocol), and SCCP (Skinny Client Control Protocol). Protocol selection should be based on various criteria including the following:
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Dennis Hartmann, CCIE No. 15651, is a Unified Communications consultant and author of Implementing Cisco Unified Communications Manager, Part 1. Dennis is also a lead instructor at Global Knowledge. Dennis was first exposed to CallManager during the CallManager 2.0 time frame when Cisco acquired Selsius. Dennis has various certifications, including the Cisco CCVP, CCSI, CCNP, CCIP, and the Microsoft MCSE. Dennis has worked for various Fortune 500 companies, including AT&T, Sprint, Merrill Lynch, KPMG, and Cabletron Systems. He lives with his wife and children in Hopewell Junction, New York.
Check out the Cisco course catalog from Global Knowledge.
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