The Cisco CallManager Express is a product under the Unified Communications Products suite of Cisco. In the past it was known as CCME (Cisco Call Manager Express) but now the new name is Cisco Unified Communications Manager Express.

It is an IP Telephony system (IP PBX) for small to medium size businesses of up to 250 IP phones capacity. Basically, a CallManager Express system is a normal Cisco Integrated Services Router (models 1800, 2800, 3800) which has the CallManager software installed on the router’s flash memory. The router hosting the callmanager system can work also as normal Internet Border router or as WAN Router connecting to other enterprise sites. The CallManager software provides call control and IP telephony functionality to internal IP phones. For connectivity to the PSTN network, voice interface cards can be installed on the CallManager router (such as voice BRI, PRI etc).

In this post we will describe three common deployment models for a CallManager Express system as it is implemented in real world enterprise environments. The three deployment models are Single Site, Multi Site with Distributed Call Processing, and Multi Site with Centralized Call Processing.

Single Site Deployment Model

This is the most common scenario and is usually found in smaller business environments. See the picture below:

Basically a single CallManager Router system is installed which usually provides also the Internet connectivity for the office. If you are a little flexible with your budget, I would recommend installing a firewall in front of the CallManager router to protect it from Internet attacks. All IP Telephony services are provided on the LAN network for internal IP Voice communication. Any call beyond the LAN uses the PSTN network. There are no telephony services provided over an IP WAN.

Characteristics and Best Practices

• Maximum of 250 IP phones can be supported.
• Arrange your internal switch to have two VLANs (one for Voice and one for Data Traffic).
• Use G.711 codec for all IP phone calls on the LAN (80kbps bandwidth per call) for best voice quality.
• You can also install a Voice Mail card on the router to offer voice mail functionality to users.
• Use appropriate Voice Interface Cards on router for PSTN connectivity.
• You can use dual router for redundancy if needed.
• Try to avoid connecting the CallManager router directly to the Internet. Use a firewall as border internet device.
• Dial Plan is simplified. If DID (Direct Inward Dialing) is required, then arrange your dial plan and internal IP phone numbering accordingly.

Multi Site with Distributed Call Processing Model

The multi site model consists of two or more independent sites, each with its own CallManager Express system installed (distributed call processing) as shown in the figure below.

All the sites are interconnected over an IP WAN which can be offered via Leased Lines, Frame Relay, ATM, MPLS Layer2/3 VPN, IPSEC VPN over the Internet etc. All sites have also local PSTN connectivity which can serve as backup to the WAN telephony connectivity or for local inbound and outbound PSTN calls. 

Characteristics and Best Practices

• PSTN Call cost savings when using the IP WAN for calls between sites.
• Bypass long distance call charges (toll bypass) by routing calls through remote site callmanager systems which are closer to the PSTN number dialed. For example, you have one site in New York and one in California. Calls from NY to California can be routed over the IP WAN towards Cal office and then get out to PSTN from the Cal office.
• No loss in functionality for IP WAN failure because there are independent Call processing units in each site.
• Recommended to install a GateKeeper (Cisco IOS gatekeeper) to provide call admission control and dial-plan resolution.
• Use G.729 or G.723 codec for IP calls over the WAN to save bandwidth.
• Use a SIP proxy if you are using SIP instead of H323.

Multi Site with Centralized Call Processing Model

This implementation scenario is suitable for an Enterprise that has a big central office with several smaller branches. One centralized CallManager system can be installed to the Central Site offering call processing and IP Telephony service to both the central site as well as to the remote small branches. The remote branches are equipped only with IP phones (no callmanager system). This is shown in the figure below:

The remote branches are connected to the central site over an IP WAN or even using IPSEC VPN over the Internet. The IP phones located to the remote sites should have IP connectivity to the Central CallManager system, where they are registered. PSTN access is offered only on the Central Site. That is, the call of a remote branch user calling a PSTN number is routed over the WAN to the Central Site and then routed out to the PSTN.

Characteristics and Best Practices

• Cost savings in hardware (only one central callmanager express)
• Easier to manage (centralized management for all IP phones).
• Disadvantage in redundancy since remote sites depend heavily on the availability of WAN lines.
• Use G.729 or G.723 for inter-site calls.
• Savings in PSTN line costs.
• Remote sites must not have many IP phones (10-20 maximum).

All the above deployment models apply also for the other Cisco IP Telephony solution, the Cisco Unified Communications Manager system which is for bigger implementations compared to the Express solution.

Before showing you how to configure a basic IP phone on Cisco CallManager Express (CCME), you need first to understand the concepts of ephone and ephone-dn.

In CCME, “ephone” (short for Ethernet Phone) refers to the physical IP phone device, and is configured with the Ethernet MAC address of the IP phone. The MAC address of the IP phone uniquely identifies the device on the network and is found on a sticker on the underside of the IP phone or from the phone’s shipping carton label.

The ephone directory number (ephone-dn) refers to the phone lines that are associated with the ephone device. The ephone-dn parameter basically configures the telephone device number. Also, the ephone-dn can use the “dual-line” option which will allow the IP phone to handle two simultaneous calls. The dual-line option also provides a way for the phone to support call waiting, conferencing, call transfer with consultation etc.

Configuration:

In the following configuration we will configure a Cisco 7960 IP phone with two directory numbers 2100 and 2200 on the first two line buttons of the telephone.

CCME#show running-config

!Tell the router that the phone firmware P00303020214.bin is located in Flash

tftp-server flash:P00303020214.bin

!Configure the IP Telephony DHCP range
ip dhcp pool Voice
network 10.1.1.0 255.255.255.0
default-router 10.1.1.1
option 150 ip 10.1.1.1

interface FastEthernet0/0
ip address 10.1.1.1 255.255.255.0

telephony-service
ip source-address 10.1.1.1
load 7960-7940 P00303020214
max-ephones 24
max-dn 24
create cnf-files

!Configure the first directory number 2100
ephone-dn 10 dual-line
number 2100

!Configure the second directory number 2200
ephone-dn 11 dual-line
number 2200

!Configure the 7960 phone and assign ephone-dn numbers to buttons 1 and 2
ephone 1
mac-address 000d.aa45.3f6e
type 7960
button 1:10 2:11

For many years now, telephony voice services for businesses and enterprises have been provided by using legacy PBX systems connected to the Public Switched Telephone Network (PSTN) using TDM connections (T1/E1 ISDN PRI lines or BRI or analog lines). This is shown on the figure below:

Newer telephony systems adopted the IP technology on the internal LAN, but they still used TDM connectivity (ISDN PRI/BRI and analog lines) to connect to the legacy PSTN network as shown below:

The newest trend is to go all-IP using SIP TRUNKING to connect your business office to the Telephony Service Provider network. A SIP Trunk allows the company to replace the traditional TDM fixed lines (PRI, BRI etc) with just a normal IP connection towards the service provider. This solution offers significant cost savings to the enterprise as you avoid costly BRI/PRI lines. Also, voice/data traffic can be converged on a single IP connection. This scenario is shown below:

The Cisco Call Manager Express product can be used as the telephony SIP trunk gateway between the local IP telephony network and the IP Telephony Service Provider. Calls from and to PSTN will be handled by a SIP PROXY server located in the Service Provider network.

 A sample Call Manager Express configuration for SIP trunking is shown below (a snippet of the complete configuration is shown):

voice service voip
   allow-connections sip to sip
   sip
       registrar server expires max 3600 min 3600
       localhost dns:mycompany.test.com

voice class codec 1
 codec preference 1 g711ulaw

!— Inbound Translation Rule
!—  for Auto Attendant pilot number “500″
voice translation-rule 1
 rule 1 /5552222100/ /500/

voice translation-profile AutoAttendant
!— Applied to the inbound dial-peers for AA
 translate called 1

!— SIP Trunk Configuration —
dial-peer voice 1 voip
 description **Incoming Call from SIP Trunk**
 translation-profile incoming AutoAttendant
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 incoming called-number .%
 dtmf-relay rtp-nte
 no vad

dial-peer voice 2 voip
 description **Outgoing Call to SIP Trunk**
  destination-pattern 9……….
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad

dial-peer voice 3 voip
 description **International Outgoing Call to SIP Trunk**
  destination-pattern 9011T
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad

!— SIP UA Configuration —
sip-ua
 authentication username 5552222100 password 075A701E1D5E415447425B
 no remote-party-id
 retry invite 2
 retry register 10
 retry options 0
 timers connect 100
 registrar dns: mycompany.test.com expires 3600
 sip-server dns: mycompany.test.com
  host-registrar
!

The Cisco Unified CallManager Express (CME) solution not only has the benefit of voice-data integration on a single platform, but offers also flexible deployment options. The Cisco CME on its basic form consists of a router on which the callmanager software is installed, plus several telephony devices. The CME router acts as a gateway between the Public Switched Telephone Network (PSTN) and your local IP telephony network. IP Phones or other legacy telephony devices can be connected on the Call Manager Express router (either directly using FXS ports, or on the local LAN switch). The figure below shows a basic small-office/medium-office CME network topology (figure is from Cisco):

The typical CME deployment above uses a single callmanager router with few legacy telephony devices (normal telephones and a Fax machine) connected directly on the router itself (on FXS ports), plus few IP Phones connected on the local LAN switch. All these phones are controlled by the CME router.

The Cisco CME software uses the following basic building blocks:
 

• Ephone: This is configured in software (using IOS commands on the router) and represents a physical telephone. The MAC address of each physical phone is configured using the ephone configuration commands.
• Directory Number: This is again a software concept that represents the line that connects a voice channel to a phone. A directory number represents a virtual voice port in the Cisco Unified CME system.

Call Manager Express Call Handling Modes

Before deploying a Call Manager Express system you must decide how the system will handle calls. There are three call handling models: PBX model, KeySwitch model or Hybrid model.

PBX Model:
This is the simplest and most popular call manager mode of operation. Each internal telephone has its own unique directory number (extension number) as shown in the diagram below.

Incoming PSTN calls are usually routed by the CME router to a central receptionist (or auto-attendant) which then delivers the calls to the appropriate requested extension number. There is also the option of having Direct Inward Dialing (DID) lines towards the PSTN which allows incoming PSTN calls to be directly routed to specific internal extensions. An example of DID is when calls coming to number 555-838-1001 will be routed directly to Extension 1001, calls coming to number 555-838-1002 will be routed to Extension 1002 etc.

It is recommended for this model that you configure directory numbers as dual-lines so that each button that appears on an IP phone can handle two concurrent calls. Dual-line directory numbers enable your configuration to support call waiting, call transfer with consultation, and three-party conferencing (G.711 only).

Keyswitch Model:
In this model there is no central receptionist telephone. Rather, all telephones have an identical configuration in which each phone is able to answer any incoming PSTN call on any line. An example is shown below:

The keyswitch model is configured by creating a set of directory numbers (Extension numbers) that correspond one-to-one with your PSTN lines. Then you configure your PSTN ports to route incoming calls to those directory numbers. When an incoming PSTN call arrives (e.g on Extension 1001), then ALL telephones will ring on line 1001. Any user can then pick-up the ringing line by just pressing the button corresponding to that line.

 Hybrid Model:
In this model, each IP phone can have both PBX and Keyswitch configurations. Each telephone can have unique extension numbers (PBX model) and also shared lines numbers (keyswitch model).

A hybrid model is shown above. Extension numbers 1001, 1002, 1003 are shared lines, and Extensions 1004, 1005, 1006 are unique private numbers for each user.