Cisco 640-461 (ICOMM) Exam Training

Explore the basics of voice communications and how voice over IP interacts with traditional phone systems, including copper cabling, tip and ring, loop start, glare, and PBX considerations.

Discover how digitized voice over long distance networks uses TDM and channelized signaling on T1/E-1, improves copper efficiency, and contrasts analog versus digital, including CAS and CCS.

Learn voice over IP terminology and key codecs like G.711 and G.729, including A-law and mu-law, and how bandwidth, T-38 fax, and DTMF tones affect IP calls.

Learn voice over IP terminology, including transcoding, DSP resources, and MOS scoring for voice quality. Explore latency and jitter, end-to-end performance, and how Cisco Unified Communications Manager uses DSP farms.

Learn how voice over IP traffic flows on a LAN in a Cisco environment, from switch to co-manager, to end-to-end endpoint calls with a path versus call control path.

Explore voice codecs in voip, explaining how analog voices convert to digital using Nyquist sampling at twice the highest frequency, yielding about 8000 samples per second and supporting packetized calls.

Explore voip sampling and quantization, using 8k samples per second and 8-bit samples, with a-law and mu-law flavors producing 64 kbps per channel.

Explore voice codecs, their bandwidth needs, and trade-offs; see how MOS, jitter, and packet loss influence codec choice, and how Cisco’s DSP calculator supports iLBC, G.729, and G.711.

Explore how VoIP traffic travels over UDP, and how RTP carries payload while RTCP provides statistics on packet loss, delay, and jitter to monitor call quality.

CUCM Express runs on router hardware as a stripped down CUCM, ideal for sites, scales to 450 phones, and is configurable via CLI or CCP web, with phone licenses separate.

Discover Cisco unified communications manager, the enterprise core of the UCM platform, bridging co-manager express to full deployments, with publisher and subscriber servers, VM, clustering, and 40,000 endpoints per cluster.

This lecture explains how phones register with the Cisco call manager, including MAC address validation, manual versus auto registration, and DHCP option 150 directing TFT configuration and firmware download.

Discover Cisco unified communications manager express, a small business phone system for 50–150 users that runs on integrated services routers and supports voip, pstn, and sip trunks.

Configure CUCM Express for the CCP app by enabling http/https and setting local authentication. Create a local user CCP with privilege 15 and enable telnet/ssh on vty lines, then save.

Launch the CCP GUI to discover and configure a Cisco router, explore unified communications features, and set up IP telephony with co-manager express options; save changes to startup configuration.

Learn basic CME configuration on the co-manager express platform, including setting max phones and directory numbers, binding telephony services to a loopback IP for CCP or web-based management.

install web gui administration on the router using the co-manager express tar; it includes telephony upgrade tools, verify the tar version matches your cmt, then extract to flash with archive.

Continue installing the web GUI by extracting the full package to router flash, review sbm and bbn firmware files, and prepare readme-driven tftp server commands for proper file serving.

Enable the CMU web GUI by starting the IP HTTP server, setting the web path, and creating a web admin user, then preview the GUI to manage extensions and phones.

Configure the Cisco IP communicator to register with co-manager express by setting a MAC-based device name, selecting the network adapter, and pointing to TFT IP servers (option 150).

Configure the Cisco ip communicator in global mode, set the mac address and type, then enable auto registration in telephony service for registration with the co-manager express router.

Configure a directory number (dn) and assign it to a phone via the GUI, creating extension 1150 with a normal ring extension, then save and verify on the phone.

Configure directory numbers in CCP by creating an extension (1150, front desk), selecting line type, and enabling features like find me follow me and presence before creating two extensions.

Configure a front desk phone in CCP by selecting the model, assigning extensions and lines, and registering the IP communicator with call manager.

Identify voice port types, including analog RJ-11 ports (FXO/FXS) for POTS and devices like fax machines, and digital PRI ports (T1/E1) with bearer and data channels.

Understand how dial peers route local and off-net calls between extensions and the PSTN, using inbound and outbound dial peers, call legs, and voice gateways.

Configure basic voice ports on a Cisco router and monitor their status with show voice port summary. Set call progress tones and station-id for analog devices.

Configure T-1 voice ports on a Cisco router by setting card type, slot, ESF framing, and DS0 group 1 with time slots 1–24; verify voice port summary.

Configure dial peers by matching a destination pattern and routing to a voice port or remote voip ip, using pots or voip peers and wildcard patterns.

Explore destination pattern wildcards in dial-peer configuration, including period, brackets, caret negation, plus, and pound, and learn how the most specific pattern wins and affects routing.

Learn to combine destination patterns and wild cards to create complex dial plans, using ranges in brackets, the t wildcard, and international formats like 9011 with a trailing pound.

Set up and verify dial peers on the voice gateway to route VOIP calls to analog lines, demonstrating destination patterns, session targets, and port mappings with a live test.

set up a voip dial-peer and dial-up on the voice gateway to route calls for extension 1150 to the remote router, matching destination patterns and using session target 192.168.151.185.

Learn to configure POTS dial peers with Cisco Configuration Professional, including creating dial-ups, destination patterns, priority, and trunk selection, and understand the router commands delivered to the device.

Learn to configure voip dial peers using CCP on a Cisco gateway, including dial peer properties, signaling protocols, codecs, and options like VAD and DTMF.

Explore how dial-peers route calls on Cisco gateways, including inbound dial-peer selection by incoming called number, answer address, and destination pattern, plus why dial-peer zero and a default dial-peer matter.

Learn how to manipulate dialed digits in pots and voip environments, including digit stripping, forward digits, prefixed digits, and the voice translation profile to apply translation rules for call routing.

Learn how class of restrictions in Cisco Unified Communications Manager Express controls calls with inbound dial peers using locks and keys, and how partitions and calling search spaces enforce routing.

Explore configuring class of restriction using dial-up entries to separate local, long distance, emergency, and toll-free calls in unified communications manager, with a custom configuration step.

Explore configuring calling search space in voice gateways by creating outbound and inbound core lists, applying them to dial peers, and restricting calls (local, long distance, emergency).

Perform a classic restriction analysis of a Cisco dial-peer setup, showing how local, emergency, and international calls through pots are controlled by inbound and outbound lists.

Explore the call handling features in manager express, including directory lookup, call park, call pickup, intercom, paging, call forwarding and transfer, blocking, cdr, single number reach, and music on hold.

Learn to configure and manage a local phone directory, populate caller ID fields from extensions, search and dial entries, and adjust directory sorting and labeling via web and CLI.

Configure call parks in Cisco configuration pro with CCP, creating a park name, slots, and a starting DN. Define timeouts and reminders to manage parked calls.

Learn how to configure general and directed call park, manage reminder and termination phases, and set timeouts and redirection rules to extensions when calls are parked.

Configure call pickup groups in the Cisco telephony features to let a dn on a pickup group allow any one person to answer calls on other phones, including extensions.

Configure intercoms by creating two speed-dial extensions, selecting whisper mode or hold and barge options, and pushing the configuration to IP phones with distinctive labeling to prevent accidental dialing.

Configure paging with Cisco configuration professional, creating East and West paging numbers 60100 and 60101. Use unicast paging for directed paging, removing multicast IPs and capping groups at 10 phones.

Explore call forwarding and transfer, routing a call from one extension to another via the user phone or administrator settings, forward all calls and transfer modes (blind, consult) per DN.

Define after-hours call blocks with co-manager express by creating blocked patterns, assign weekly and holiday schedules, and apply per-phone or pin-based exemptions using Cisco configuration professional.

Configure call accounting to capture CDRs and send them to the local router log and assist log, and assign accounting numbers.

Configure single number reach at the extension level with Cisco configuration professional, defining remote numbers, ring delay, forward unanswered calls, timeouts, and mobility.

Explore how quality of service prioritizes voice and data by using classification and marking, policing and shaping, congestion management, and congestion avoidance to ensure bandwidth, delay, and packet loss control.

Explore the network requirements for a functional voice over IP implementation, focusing on end-to-end delay, jitter, and packet loss, and how latency across the PSTN impacts call quality.

Analyze jitter, delay, and packet loss in VoIP networks, and how Cisco jitter buffers and QoS mitigate variability to ensure reliable voice across WAN links.

Explore qos models: best effort, integrated services with RSVP bandwidth reservation, and differentiated services that classify and mark traffic using DSCP in layer 3 and layer 2 class of service.

Explore how routers manage congestion with queuing, from first in, first out to weighted fair queuing, class based weighted fair queuing, and low latency queueing, prioritizing VoIP.

Learn how auto qos simplifies voice traffic configuration on routers and switches, with voip classification for Cisco phone or softphone on interface gigabit 0/25.

Learn the Cisco unified communications manager platform (CUCM), the enterprise VoIP PBX, and its publisher and subscriber cluster architecture with web-based configuration.

Explore the Cisco Unified Communications Manager admin web interface to configure phones, licenses, time zones, DHCP and LDAP information, and enterprise and service parameters.

Explore call routing in the CUCM admin interface, configuring route patterns, route groups, translation patterns, partitions, and calling search spaces, including directory numbers, call park, and call pickup.

Master OS administration through the Cisco unified OS web interface, managing IP settings, SSL certificates, upgrades, alarms, and hardware with separate platform and administration credentials.

Learn to administer the operating system from the web interface using Cisco unified serviceability, enabling and troubleshooting services across a cluster, including call manager, service activation, and dialed number analyzer.

Explore CUCM's command line interface, its relationship to the web interface, and practical tips using putty, tab completion, and utilities like system restart and show disk usage.

Register a Cisco VoIP phone by powering it from the switch, booting its Java-based firmware, obtaining IP via DHCP and TFTP, then registering with Unified Communications Manager for line settings.

Explore power over ethernet for IP phones, comparing power bricks, injectors, and patch panels. Understand when sidecars require direct power and why QoS and voice VLANs matter for VoIP.

Learn how a voice vlan isolates voice traffic, enables QoS policies, and uses vlan tagging to prioritize IP phone data on Cisco switches.

Configure dhcp for IP phones on a Cisco Unified Communications Manager cluster, using option 150 to point to TFT servers and deliver registration and operating configuration.

Configure TFT services and option 150, then provision the Cisco IP communicator in the CUCM administration web interface to register with the co-manager publisher.

Discover bulk administration for Cisco Unified Communications Manager by gathering MAC addresses from phone menus, labels, or box barcodes and importing them in bulk via the CSV-based bulk administration tool.

Learn to use the bulk administration tool to prepare a bat file for importing phones into Call Manager, including MAC addresses, directory numbers, and optional fields like device trust mode.

Learn to bulk administer phones by uploading a CSV, selecting insert phones with specific details, validating the template, and running import jobs with dummy MAC addresses.