Cisco VoIP CME Labs by Michael T. Durham

Size: px
Start display at page:

Download "Cisco VoIP CME Labs by Michael T. Durham"

Transcription

1 Cisco VoIP CME Labs by Michael T. Durham Welcome to NetCertLabs CCNA Voice Lab series. In this lab we will be bringing a little sound to our callers on hold. By having MoH (Music on Hold) enabled on your CME router and a caller is placed on hold, they are assured that they are still connected to the system. Music on hold helps keep callers from hanging up during long waits and provides an outlet to tell your clients the latest news about the company or the current specials. Based on AT&T research: o 70% of calls are placed "On-Hold for 30 seconds or longer, leaving a caller on "Silence Hold" results in a staggering 75% call abandonment rate! Majority of these callers will not call back again. o 80% of callers with silence on-hold hang up within 1 minute. o Callers with Music-on-Hold stay on-line up to five minutes longer o 30% of callers purchased additional products or services as a result of something they heard on-hold o 25% of callers make a purchase based on an on-hold sales suggestion Based on a CNN Survey: o The average person spends 100 hours per year on-hold" o "Without music or messages, 70% of those on-hold will hang up and 35% won't call back" Cellular Marketing Magazine: o "Over 90% of callers prefer on-hold messages over silence" Telemarketing Magazine: o "Surveys show that 25% to 40% of callers make purchases based on information they heard on-hold" Cisco supports two forms of music on hold streams, an audio file on the flash drive or a live streaming source via an FXO or E&M voice card. There are a few restrictions you need to be aware of when setting up MoH. They are: Phones receiving MOH in a system using G.729 require transcoding between G.711 and G.729. For information about transcoding. IP phones do not support multicast at 224.x.x.x addresses. Cisco Unified CME 4.0 and later versions support MOH for internal calls only if the multicast moh command is used to enable the flow of packets to the subnet on which the phones are located. Internal extensions that are connected through a Cisco VG224 Analog Voice Gateway or through a WAN (remote extensions) do not hear MOH on internal calls. Multicast MOH is not supported on a phone if the phone is configured with the mtp command or the paging-dn command with the unicast keyword. More information on MoH can be found at: Equipment used in this lab: 1 Cisco 2610XM or 28x1 router 1 VIC2-2FXO, VIC2-4FXO, EM-HDA-3FXS/4FXO, EM-HDA-6FXO, and EM2-HDA-4FXO, VIC-2E&M, VIC2-2E&M * 1 MOD-SC adapter* 1 Cisco 3524 layer three switch (SMI or EMI) PoE switch preferred 1 Personal Computer (desktop or laptop) 2 Cisco 79x0 IP Phone 5 Cat 5 Ethernet Patch Cables *These items are only needed if you want to have a live stream for your music on hold source

2 1 2 ABC 4 5 GHI JKL 7 8 PQRS T UV 0 OPER 3 DEF 6 MNO 9 WXYZ # CISCO IP PHONE 7941 SERIES -? ABC 4 5 GHI JKL 7 8 PQRS T UV 0 OPER 3 DEF 6 MNO 9 WXYZ # CISCO IP PHONE 7941 S ERI ES -? + Cisco VoIP CME Lab 14 - Music on Hold Below is the network diagram for the network that we will be setting up in the CME (Call Manager Express) VoIP lab series. Cisco Internet DSL/Cable Provided Router VLAN 1 VLAN x (dhcp) DSL/Cable Modem Cisco 2610XM or 28x1 fa0/ x (dhcp) fa0/ fa0/ * x x (dhcp) (dhcp) VLAN 2 * This lab assumes that you have configured your lab by following NetCertLabs VoIP CME labs 1 through 13 with all interfaces configured, IP addresses assigned, and other configurations completed. Depending on which version of CME you are running determines the features available to you. Most versions of CME support only one audio stream source. However, CME 8.0 and above support an enhanced MoH system that supports several audio sources and you can select groups to receive a selected audio stream. For example, internal users would hear one source of MoH while external callers would hear a different one. MOH is an audio stream that is played to PSTN and VoIP G.711 or G.729 callers who are placed on hold by phones in a Cisco Unified CME system. The table below shows the three ways you can configure a source of music on hold. When the phone receiving MOH is part of a system that uses a G.729 codec, G.711 MOH must be translated to G.729. Note that because of compression, MOH using G.729 is of significantly lower fidelity than MOH using G.711. Audio Source Description Flash memory No external audio input is required. Live feed Live feed and flash memory The multicast audio stream has minimal delay for local IP phones. The MOH stream for PSTN callers is delayed by a few seconds. If the live feed audio input fails, callers on hold hear silence. The live feed stream has a few seconds of delay for both PSTN and local IP phone callers. The flash MOH acts as backup for the live-feed MoH. We recommend this option if you want live-feed because it provides guaranteed MOH if the live-feed input is not found or fails. Music on hold requires either a live audio stream, an audio file in.au or.wav format or both. You can record prompts for Cisco Call Manager Express Auto Attendant (AA) on any computer using Microsoft Sound Recorder or any other audio program that can save the file as a.wav file in CCITT ( -law) 8kHz, 8-bit, mono format There is no file size limitation other than the amount of free space on your flash drive. The first step is to record a music on hold file.

3 Once you have your MoH audio file, you need to upload it to your flash: drive on your Cisco router. LAB_2851#copy tftp flash Address or name of remote host []? Source filename []? hold.wav Destination filename [hold.wav]? Accessing tftp:// /hold.wav... Loading hold.wav from (via GigabitEthernet0/1):!!!!! [OK bytes] bytes copied in secs ( bytes/sec) To enable MoH we need to enter the telephony-service mode and add some configurations. LAB_2851(config)#telephony-service The next command sets the file we uploaded above to be the music on hold file. LAB_2851(config-telephony)#moh hold.wav If you specify a file with this command and later want to use a different file, you must disable use of the first file with the no moh command before configuring the second file The multicast moh command specifies that the audio stream is to be used for multicast and also for music on hold. This command is required to use music on hold for internal calls and it must be configured after music on hold is enabled with the moh command. Cisco recommends that you configure multicast MoH audio sources to use IP addresses in the range to We recommend port 2000 because it is already used for normal RTP media transmissions between IP phones and the router. LAB_2851(config-telephony)#multicast moh port 2000 For more on multicasting Moh see: Should you have music on hold for internal calls on hold and not external callers, (or visa versa), check the codecs and make sure all of your codecs are set to G711ulaw. If you have DSP s installed in your system, check your transcoding settings. Now that you have a working music on hold, let s add a live audio stream to our system. You need to pieces of hardware to make this happen, an E&M or FXO adapter as listed above. You will also need an adapter the connects the RJ-45 connector to your audio source, usually an 3.5mm (1/8 inch) jack. You can make one of these adapters yourself by buying the 3.5mm stereo male jack and an RJ-45 female jack at an electronics store. Connect on wire to the tip of the 3.5mm jack and one to the back (ground) connector. If you are using and E&M card the two wires connect to the RJ-45 female connector s ping 3 and 6. It does not matter which wire goes to

4 which pin. And if you are using and FXO card, then you would use and RJ-11 female jack and connect the wires to pins 3 and 4. To configure music on hold using a live feed, you need to establish a voice port and dial peer for the call and also create a "dummy" ephone-dn. The ephone-dn must have a phone or extension number assigned to it so that it can make and receive calls, but the number is never assigned to a physical phone. Only one live music on hold feed is supported per system. The E&M voice interface card (VIC) has a built-in audio transformer that provides appropriate electrical isolation for the external audio source. An audio connection on an E&M port does not require loop-current. Set the following commands if you are using an E&M card. LAB_2851(config)#voice-port 1/0/0 LAB_2851(config-voiceport)#description MOH Live Feed LAB_2851(config-voiceport)#timeouts call-disconnect 1 LAB_2851(config-voiceport)#auto-cut-through LAB_2851(config-voiceport)#operation 4-wire LAB_2851(config-voiceport)# signal immediate If you are connection your live stream through and FXO port, you can directly connect a live-feed source to an FXO port if the signal loop-start live-feed command is configured on the voice port. LAB_2851(config)#voice-port 1/0/0 LAB_2851(config-voiceport)#description MoH Live Feed LAB_2851(config-voiceport)#timeouts call-disconnect 1 LAB_2851(config-voiceport)# signal loop-start live-feed You can adjust the sound level with the input gain decibels command. The range is -6 through 14. LAB_2851(config-voiceport)#input gain decibels 3 Enable the voice-port with the no shutdown command. LAB_2851(config-voiceport)#no shutdown Next we configure a dial-peer for the live audio stream. LAB_2851(config)#dial peer voice 500 pots LAB_2851(config-dialpeer)#description Live Stream MoH LAB_2851(config-dialpeer)#destination-pattern 7000 LAB_2851(config-dialpeer)#port 1/0/0 Now let s create the ephone-dn for the live audio stream. LAB_2851(config)#ephone-dn 70 LAB_2851(config-ephone-dn)#number 7001 This number is not assigned to any phone; it is only used to make and receive calls that contain an audio stream to be used for MoH. To specify that this ephone-dn is to be used for an incoming or outgoing call that is the source for an MoH stream, use the moh out-call # command. LAB_2851(config-ephone-dn)#moh out-call 7000

5 You can use the command moh out-call 7000 ip port 200 route however, you will not have failover to the file on the flash drive should you lose the live stream. For additional feature commands such as music on hold groups and configuring buffer size, see the link to Cisco s website at the top of this document. After you have setup and tested this lab, please blog your experience on our blog site at: Thank You,

Integrating Cisco CallManager and Cisco SRST to Use Cisco SRST as a Multicast MOH Resource

Integrating Cisco CallManager and Cisco SRST to Use Cisco SRST as a Multicast MOH Resource Integrating Cisco CallManager and Cisco SRST to Use Cisco SRST as a Multicast MOH Resource Americas Headquarters Cisco Systems, Inc. 170 West Tasman Drive San Jose, CA 95134-1706 USA http://www.cisco.com

More information

Cisco VoIP CME QoS Labs by Michael T. Durham

Cisco VoIP CME QoS Labs by Michael T. Durham Cisco VoIP CME QoS Labs by Michael T. Durham Welcome to NetCertLabs CCNA Voice Lab series. In this set of labs we will be working with the QoS (Quality of Service). A communications network forms the backbone

More information

- Basic Voice over IP -

- Basic Voice over IP - 1 Voice over IP (VoIP) - Basic Voice over IP - Voice over IP (VoIP) is a digital form of transport for voice transmissions, replacing analog phone systems. The benefits of VoIP are considerable: Better

More information

Direct Inward Dial Digit Translation Service

Direct Inward Dial Digit Translation Service Direct Inward Dial Digit Translation Service In Cisco CME 3.2.3 and later versions, a Tcl script is available to provide digit translation for Direct Inward Dial (DID) calls when the DID digits provided

More information

IMPLEMENTING CISCO IOS TELEPHONY AND UNIFIED COMMUNICATIONS EXPRESS (IITUCX)

IMPLEMENTING CISCO IOS TELEPHONY AND UNIFIED COMMUNICATIONS EXPRESS (IITUCX) Temario IMPLEMENTING CISCO IOS TELEPHONY AND UNIFIED COMMUNICATIONS EXPRESS (IITUCX) This course is designed to be the primary training for Cisco Unified Communications Manager Express and Cisco Unity

More information

Implementing Cisco IOS Telephony and Unified Communications Express (IITUCX)

Implementing Cisco IOS Telephony and Unified Communications Express (IITUCX) Implementing Cisco IOS Telephony and Unified Communications Express (IITUCX) Course Objectives Explain the benefits and components of a Cisco Unified Communications system Describe how traditional telephony

More information

IP PBX. SD Card Slot. FXO Ports. PBX WAN port. FXO Ports LED, RED means online

IP PBX. SD Card Slot. FXO Ports. PBX WAN port. FXO Ports LED, RED means online 1 IP PBX SD Card Slot FXO Ports PBX LAN port PBX WAN port FXO Ports LED, RED means online 2 Connect the IP PBX to Your LAN Internet PSTN Router Ethernet Switch FXO Ports 3 Access the PBX s WEB GUI The

More information

Optimizing Converged Cisco Networks (ONT)

Optimizing Converged Cisco Networks (ONT) Optimizing Converged Cisco Networks (ONT) Module 2: Cisco VoIP Implementations (Deploy) Calculating Bandwidth Requirements for VoIP Objectives Describe factors influencing encapsulation overhead and bandwidth

More information

Implementing Cisco IOS Telephony and Unified Communications Express (IITUCX)

Implementing Cisco IOS Telephony and Unified Communications Express (IITUCX) Implementing Cisco IOS Telephony and Unified Communications Express (IITUCX) Who should attend The primary audience for this course is as follows: Network administrators Network engineers Systems engineers

More information

EarthLink Business SIP Trunking. Toshiba IPedge Customer Configuration Guide

EarthLink Business SIP Trunking. Toshiba IPedge Customer Configuration Guide EarthLink Business SIP Trunking Toshiba IPedge Customer Configuration Guide Publication History First Release: Version 1.0 August 30, 2011 CHANGE HISTORY Version Date Change Details Changed By 1.0 8/30/2011

More information

CONNECTING PHONES, FAXS & DEVICES TO TALKSWITCH

CONNECTING PHONES, FAXS & DEVICES TO TALKSWITCH TALKSWITCH QUICK GUIDE CONNECTING PHONES, FAXS & DEVICES TO TALKSWITCH CONNECTING PHONES, FAXES & DEVICES TO TALKSWITCH CT.TS005.504.EN - 03 TalkSwitch Back Panel TalkSwitch 48-CA/ 48-CVA shown here. Model

More information

640-460. IIUC Implementing Cisco IOS Unified Communications (IIUC) Version: Demo. Page <<1/9>>

640-460. IIUC Implementing Cisco IOS Unified Communications (IIUC) Version: Demo. Page <<1/9>> 640-460 IIUC Implementing Cisco IOS Unified Communications (IIUC) Version: Demo Page 1. You are CCNA VOICE associate in XXXX.com. You need configure a voice port that will allow the gateway to

More information

CVOICE Exam Topics Cisco Voice over IP Exam #642-432 7/14/2005

CVOICE Exam Topics Cisco Voice over IP Exam #642-432 7/14/2005 CVOICE Exam Topics Cisco Voice over IP Exam #642-432 7/14/2005 From the Cisco CVOICE 642-432 Exam Topics Voice over IP Technologies Describe the similarities and differences between PSTN and VoIP including

More information

Session Title: Exploring Packet Tracer v5.3 IP Telephony & CME. Scenario

Session Title: Exploring Packet Tracer v5.3 IP Telephony & CME. Scenario Session Title: Exploring Packet Tracer v5.3 IP Telephony & CME Scenario With the scheduled release of Packet Tracer v5.3 in the near future, this case study is designed to provide you with an insight into

More information

640-460 - Implementing Cisco IOS Unified Communications (IIUC)

640-460 - Implementing Cisco IOS Unified Communications (IIUC) 640-460 - Implementing Cisco IOS Unified Communications (IIUC) Course Introduction Course Introduction Module 1 - Cisco Unified Communications System Introduction Cisco Unified Communications System Introduction

More information

User Manual. SIP Analog Telephone Adaptor SIP-GW2. Sedna Advanced Electronics Ltd. www.sednacomputer.com

User Manual. SIP Analog Telephone Adaptor SIP-GW2. Sedna Advanced Electronics Ltd. www.sednacomputer.com User Manual SIP-GW2 SIP Analog Telephone Adaptor Sedna Advanced Electronics Ltd. www.sednacomputer.com Table of Contents 1. WELCOME... 3 2. INSTALLATION... 3 3. WHAT IS INCLUDED IN THE PACKAGE... 5 3.1

More information

Cisco Networks (ONT) 2006 Cisco Systems, Inc. All rights reserved.

Cisco Networks (ONT) 2006 Cisco Systems, Inc. All rights reserved. Optimizing Converged Cisco Networks (ONT) reserved. Lesson 2.4: Calculating Bandwidth Requirements for VoIP reserved. Objectives Describe factors influencing encapsulation overhead and bandwidth requirements

More information

642-436 Q&A. DEMO Version

642-436 Q&A. DEMO Version Cisco Voice over IP (CVOICE) Q&A DEMO Version Copyright (c) 2010 Chinatag LLC. All rights reserved. Important Note Please Read Carefully For demonstration purpose only, this free version Chinatag study

More information

EarthLink Business SIP Trunking. Asterisk 11.2 IP PBX Customer Configuration Guide

EarthLink Business SIP Trunking. Asterisk 11.2 IP PBX Customer Configuration Guide EarthLink Business SIP Trunking Asterisk 11.2 IP PBX Customer Configuration Guide Publication History First Release: Version 1.0 August 30, 2011 CHANGE HISTORY Version Date Change Details Changed By 1.0

More information

Table of Contents. Cisco Mapping Outbound VoIP Calls to Specific Digital Voice Ports

Table of Contents. Cisco Mapping Outbound VoIP Calls to Specific Digital Voice Ports Table of Contents Mapping Outbound VoIP Calls to Specific Digital Voice Ports...1 Introduction...1 Before You Begin...1 Conventions...1 Prerequisites...1 Components Used...1 Configure...2 Network Diagram...2

More information

VoIP Configuration Examples

VoIP Configuration Examples APPENDIX C This section uses four different scenarios to demonstrate how to configure Voice over IP (VoIP). The actual VoIP configuration procedure depends on the topology of your voice network. The following

More information

LevelOne VOI-9000. H.323 VoIP Gatekeeper. User Manual

LevelOne VOI-9000. H.323 VoIP Gatekeeper. User Manual LevelOne VOI-9000 H.323 VoIP Gatekeeper User Manual Quick Guide Step 1: Broadband (ADSL/Cable Modem) Connections For VOI-9000 A. Connect VOI-9000 RJ45 LAN port to Router/ADSL as one of the following connections.

More information

SL1100 Installation Instructions. This Installation Packet is Organized into Five Chapters:

SL1100 Installation Instructions. This Installation Packet is Organized into Five Chapters: Thank you for choosing NEC SL1100 Distributors! This Installation Packet is Organized into Five Chapters: BLUE: Hardware Installation Overview (Block Wiring, Music On-Hold, Paging, Door Box & Relays) GREEN:

More information

FortiVoice. Version 7.00 Start Guide

FortiVoice. Version 7.00 Start Guide FortiVoice Version 7.00 Start Guide FortiVoice Version 7.00 Start Guide Revision 2 18 October 2011 Copyright 2011 Fortinet, Inc. All rights reserved. Contents and terms are subject to change by Fortinet

More information

Network Connection Considerations for Microsoft Response Point 1.0 Service Pack 2

Network Connection Considerations for Microsoft Response Point 1.0 Service Pack 2 Network Connection Considerations for Microsoft Response Point 1.0 Service Pack 2 Updated: February 2009 Microsoft Response Point is a small-business phone solution that is designed to be easy to use and

More information

Dial Peer Configuration Examples

Dial Peer Configuration Examples Dial Peer Configuration Examples This appendix contains a series of configuration examples featuring the minimum required components and critical Cisco IOS command lines extracted from voice gateway configuration

More information

IP Telephony. User Guide. System SPA9000. Model No. Voice

IP Telephony. User Guide. System SPA9000. Model No. Voice IP Telephony System User Guide Voice Model No. SPA9000 Copyright and Trademarks Specifications are subject to change without notice. Linksys is a registered trademark or trademark of Cisco Systems, Inc.

More information

Implementing Cisco Voice Communications and QoS

Implementing Cisco Voice Communications and QoS Implementing Cisco Voice Communications and QoS Course CVOICE v8.0; 5 Days, Instructor-led Course Description Implementing Cisco Voice Communications and QoS (CVOICE) v8.0 teaches learners about voice

More information

Configuring Voice over IP

Configuring Voice over IP CHAPTER 4 This chapter explains how to configure voice interfaces and ports, which convert telephone voice signals for transmission over an IP network. This chapter presents the following major topics:

More information

My Account Quick Start

My Account Quick Start My Account Quick Start for Verizon Business Digital Voice Service Guide for Office System Administrators Accessing My Account Phone Assignment Defining the User Site Services Auto Attendant Voice Portal

More information

2100 Series VoIP Phone

2100 Series VoIP Phone 2100 Series VoIP Phone Installation and Operations Manual Made in the USA 3 Year Warranty N56 W24720 N. Corporate Circle Sussex, WI 53089 RP8500SIP 800-451-1460 262-246-4828 (fax) Ver. 4 www.rathmicrotech.com

More information

EarthLink Business SIP Trunking. ININ IC3 IP PBX Customer Configuration Guide

EarthLink Business SIP Trunking. ININ IC3 IP PBX Customer Configuration Guide EarthLink Business SIP Trunking ININ IC3 IP PBX Customer Configuration Guide Publication History First Release: Version 1.0 August 30, 2011 CHANGE HISTORY Version Date Change Details Changed By 1.0 8/30/2011

More information

Technical Configuration Notes

Technical Configuration Notes MITEL SIPCoE Technical Configuration Notes Configure Inn-Phone SIP Phone for use with MCD SIP CoE NOTICE The information contained in this document is believed to be accurate in all respects but is not

More information

Cisco Unified Communications 500 Series Model UC 560

Cisco Unified Communications 500 Series Model UC 560 Quick Start Guide Cisco Small Business Pro Cisco Unified Communications 500 Series Model UC 560 Package Contents Cisco Unified Communications 500 Series Model UC 560 4 rubber mounting feet for desktop

More information

GW400 VoIP Gateway. User s Guide

GW400 VoIP Gateway. User s Guide GW400 VoIP Gateway User s Guide P/N: 956YD30001 Copyright 2006. All Rights Reserved. Document Version: 1.0 All trademarks and trade names are the properties of their respective owners. i Table of Contents

More information

210-060. Implementing Cisco Collaboration Devices v1.0. Version: Demo. Page <<1/10>>

210-060. Implementing Cisco Collaboration Devices v1.0. Version: Demo. Page <<1/10>> 210-060 Implementing Cisco Collaboration Devices v1.0 Version: Demo Page 1. Which two technologies comprise a Cisco Presence deployment? (Choose two.) A. Cisco Unified Presence Server B. Cisco

More information

Frequently Asked Questions about Integrated Access

Frequently Asked Questions about Integrated Access Frequently Asked Questions about Integrated Access Phone Service How are local, long distance, and international calls defined? Local access transport areas (LATAs) are geographical boundaries set by the

More information

EarthLink Business SIP Trunking. Avaya IPO IP PBX Customer Configuration Guide

EarthLink Business SIP Trunking. Avaya IPO IP PBX Customer Configuration Guide EarthLink Business SIP Trunking Avaya IPO IP PBX Customer Configuration Guide Publication History First Release: Version 1.0 August 30, 2011 CHANGE HISTORY Version Date Change Details Changed By 1.0 8/30/2011

More information

Quick set-up instructions for. The Avois AV-3500 IP Phone

Quick set-up instructions for. The Avois AV-3500 IP Phone Solwise Ltd. Quick set-up instructions for The Avois AV-3500 IP Phone www.solwiseforum.co.uk The Solwise Forum is designed to be the first port-of-call for technical support and sales advice for the whole

More information

Cisco Unified CME Basic Automatic Call Distribution and Auto-Attendant Service

Cisco Unified CME Basic Automatic Call Distribution and Auto-Attendant Service Cisco Unified CME Basic Automatic Call Distribution and Auto-Attendant Service Revised: January 3, 2013 Note Prior to version 4.1, the name of the product was Cisco CallManager Express. Basic automatic

More information

VoIP 110R/200R/422R/404R/440R. User s Guide

VoIP 110R/200R/422R/404R/440R. User s Guide VoIP 110R/200R/422R/404R/440R User s Guide Trademarks Contents are subject to revise without prior notice. All trademarks belong to their respective owners. FCC Warning This equipment has been tested and

More information

Voice Gateway with Router

Voice Gateway with Router Voice User Guide Model No. SPA3102 Copyright and Trademarks Specifications are subject to change without notice. Linksys is a registered trademark or trademark of Cisco Systems, Inc. and/or its affiliates

More information

H.323 / SIP VoIP Gateway VIP GW. Quick Installation Guide

H.323 / SIP VoIP Gateway VIP GW. Quick Installation Guide H.323 / SIP VoIP Gateway VIP GW Quick Installation Guide Overview This quick installation guide describes the objectives; organization and basic installation of the PLANET VIP-281/VIP-480/VIP-880/VIP-1680/VIP-2480

More information

Music On Hold and Paging

Music On Hold and Paging Music On Hold and Paging 24 Objectives When you finish this module, you will be able to: Describe Music On Hold (MOH). Program analog and digital MOH. Download a.wav file for Embedded MOH. Describe Loudspeaker

More information

Cisco Small Business Unified Communications 300 Series

Cisco Small Business Unified Communications 300 Series Cisco Small Business Unified Communications 300 Series Feature Reference Guide January 2011 Introduction The Cisco Small Business Unified Communications 300 Series is a cost-effective, fully featured unified

More information

IP Telephony v1.0 Scope and Sequence. Cisco Networking Academy Program

IP Telephony v1.0 Scope and Sequence. Cisco Networking Academy Program IP Telephony v1.0 Scope and Sequence Cisco Networking Academy Program Table of Content COURSE OVERVIEW...4 Course Description...4 Course Objectives...4 Target Audience...5 Prerequisites...5 Lab Requirements...5

More information

Enterprise VoIP. Silvano Gai. ftp://ftpeng.cisco.com/sgai/t2000voip.pdf. Cisco Systems, USA Politecnico di Torino, IT sgai@cisco.com.

Enterprise VoIP. Silvano Gai. ftp://ftpeng.cisco.com/sgai/t2000voip.pdf. Cisco Systems, USA Politecnico di Torino, IT sgai@cisco.com. Enterprise Vo Terena 2000 ftp://ftpeng.cisco.com/sgai/t2000voip.pdf Silvano Gai Cisco Systems, USA Politecnico di Torino, IT sgai@cisco.com Terena 2000 1 Compass Motivation for Vo Voice over in the Enterprise

More information

User Manual 821121-ATA-PAK

User Manual 821121-ATA-PAK User Manual 821121-ATA-PAK IMPORTANT SAFETY INSTRUCTIONS When using your telephone equipment, basic safety precautions should always be followed to reduce the risk of fire, electric shock and injury to

More information

This topic describes dial peers and their applications.

This topic describes dial peers and their applications. Dial Peers What is Dial Peer? This topic describes dial peers and their applications. What is a Dial Peer? A dial peer is an addressable call endpoint. Dial peers establish logical connections, called

More information

3.1 Connecting to a Router and Basic Configuration

3.1 Connecting to a Router and Basic Configuration 3.1 Connecting to a Router and Basic Configuration Objective This lab will focus on the ability to connect a PC to a router in order to establish a console session and observe the user interface. A console

More information

DECT Gigaset N510 IP PRO

DECT Gigaset N510 IP PRO DECT Gigaset N510 IP PRO SUMMARY Summary... 1 1. System Features of the DECT Gigaset N510 IP Pro... 2 Base Station Gigaset N510 IP Pro... 2 Environmental Requirements... 3 The base should be placed in

More information

IP Telephony Deployment Models

IP Telephony Deployment Models CHAPTER 2 Sections in this chapter address the following topics: Single Site, page 2-1 Multisite Implementation with Distributed Call Processing, page 2-3 Design Considerations for Section 508 Conformance,

More information

NetComm V90 VoIP Phone Quick Start Guide Draft Release 0.1

NetComm V90 VoIP Phone Quick Start Guide Draft Release 0.1 NetComm V90 VoIP Phone Quick Start Guide Draft Release 0.1 Copyright NetComm Ltd Overview NetComm V90 SIP VoIP Phone User Guide Table of Contents Overview... 3 V90 VoIP Phone Specification...4 Shipping

More information

P-2024. Quick Start Guide. VoIP Analog Telephone Adaptor DEFAULT LOGIN. IP Address http://192.168.5.1 Password 1234. Version 3.60 7/2007 Edition 1

P-2024. Quick Start Guide. VoIP Analog Telephone Adaptor DEFAULT LOGIN. IP Address http://192.168.5.1 Password 1234. Version 3.60 7/2007 Edition 1 P-2024 VoIP Analog Telephone Adaptor Quick Start Guide Version 3.60 7/2007 Edition 1 DEFAULT LOGIN IP Address http://192.168.5.1 Password 1234 Copyright 2007. All rights reserved. Overview Use your P-2024

More information

Internet Basics Thursday, November 20, 2008

Internet Basics Thursday, November 20, 2008 Internet Basics Thursday, November 20, 2008 Welcome to Internet Basics, the first section of the Basic Networking course of the online Allworx Reseller technical training. This course provides general

More information

CVOICE - Cisco Voice Over IP

CVOICE - Cisco Voice Over IP CVOICE - Cisco Voice Over IP Table of Contents Introduction Audience At Course Completion Prerequisites Applicable Products Program Contents Course Outline Introduction This five-day course covers the

More information

Configuring WAN Failover with a Cisco 881 Router and an AirLink ES440

Configuring WAN Failover with a Cisco 881 Router and an AirLink ES440 Configuring WAN Failover with a Cisco 881 Router and an AirLink ES440 When the AirLink ES440 is combined with a third-party router, the combined solution supports business continuity by providing primary

More information

EarthLink Business SIP Trunking. Cisco Call Manager and Cisco CUBE Customer Configuration Guide

EarthLink Business SIP Trunking. Cisco Call Manager and Cisco CUBE Customer Configuration Guide EarthLink Business SIP Trunking Cisco Call Manager and Cisco CUBE Customer Configuration Guide Publication History First Release: Version 2.0 April 20, 2012 CHANGE HISTORY Version Date Change Details Changed

More information

Introduction to VoIP Technology

Introduction to VoIP Technology Lesson 1 Abstract Introduction to VoIP Technology 2012. 01. 06. This first lesson of contains the basic knowledge about the terms and processes concerning the Voice over IP technology. The main goal of

More information

Your Technical Point of Contact s Role

Your Technical Point of Contact s Role 3 Your Technical Point of Contact s Role The TPOC will be the overall central point of contact for all issues related to AT&T Managed Internet Service with VoIP (MIS with VoIP) at your company. At a minimum,

More information

Curso de Telefonía IP para el MTC. Sesión 1 Introducción. Mg. Antonio Ocampo Zúñiga

Curso de Telefonía IP para el MTC. Sesión 1 Introducción. Mg. Antonio Ocampo Zúñiga Curso de Telefonía IP para el MTC Sesión 1 Introducción Mg. Antonio Ocampo Zúñiga Conceptos Generales VoIP Essentials Family of technologies Carries voice calls over an IP network VoIP services convert

More information

Getting Started KX-TDA5480

Getting Started KX-TDA5480 4-Channel VoIP Gateway Card Getting Started KX-TDA5480 Model KX-TDA0484 Thank you for purchasing the Panasonic 4-Channel VoIP Gateway Card, KX-TDA5480/KX-TDA0484. Please read this manual carefully before

More information

IP-PBX Quick Start Guide

IP-PBX Quick Start Guide IP-PBX Quick Start Guide Introduce... 3 Configure and set up the IP-PBX... 4 How to change the IP address... 7 Set up extensions and make internal calls... 8 How to make calls via the FXO port... 10 How

More information

Analog Telephone Adapter Network settings via Keypad commands:

Analog Telephone Adapter Network settings via Keypad commands: Analog Telephone Adapter Network settings via Keypad commands: The ATA series phone adapters (VIP-156/VIP-156PE/VIP-157/VIP-157S) support telephone keypad configurations, please connect analog telephone

More information

Cisco Unified Communications 500 Series Model 540 for Small Business

Cisco Unified Communications 500 Series Model 540 for Small Business Cisco Unified Communications 500 Series Model 540 for Small Business Reference Guide 2009 Cisco Systems, Inc. All rights reserved. This document is Cisco Public Information. Contents Product Overview 3

More information

Lab 2.1 Configure CME using the CLI and Cisco IP Communicator

Lab 2.1 Configure CME using the CLI and Cisco IP Communicator Lab 2.1 Configure CME using the CLI and Cisco IP Communicator Learning Objectives Configure Cisco Unified Call Manager Express (CME) Install Cisco IP Communicator (CIPC) on a host Verify CME and CIPC Operation

More information

Quick & Easy Set-Up of Packet8 Internet Phone Service

Quick & Easy Set-Up of Packet8 Internet Phone Service For the Way You Live & Work Quick & Easy Set-Up of Packet8 Internet Phone Service Welcome to Packet8 Internet Phone Service. Soon, you ll be able to make all your calls over the Internet and save a bundle

More information

VoIP ATA series (ATA171plus, ATA172plus, ATA-171, ATA-172, ATA-171M, ATA-171P)

VoIP ATA series (ATA171plus, ATA172plus, ATA-171, ATA-172, ATA-171M, ATA-171P) ATA Web User Guide VoIP ATA series (ATA171plus, ATA172plus, ATA-171, ATA-172, ATA-171M, ATA-171P) User Guide Released Date : January-2012 Firmware Version : V.300 1. Introduction... 4 2. Hardware Overview...

More information

Using a Sierra Wireless AirLink Raven X or Raven-E with a Cisco Router Application Note

Using a Sierra Wireless AirLink Raven X or Raven-E with a Cisco Router Application Note Using a Sierra Wireless AirLink Raven X or Raven-E with a Application Note Cisco routers deliver the performance, availability, and reliability required for scaling mission-critical business applications

More information

Direct IP Calls. Quick IP Call Mode

Direct IP Calls. Quick IP Call Mode Unicorn3112 Tips Direct IP Calls...1 Quick IP Call Mode...1 PSTN Pass Through...2 VoIP-to-PSTN Calls...2 PSTN-to-VoIP Calls...3 Route Calls to PSTN...4 Forward Calls to PSTN...4 Forward Calls to VoIP...4

More information

AP200 VoIP Gateway Series Design Features & Concept. 2002. 3.5 AddPac R&D Center

AP200 VoIP Gateway Series Design Features & Concept. 2002. 3.5 AddPac R&D Center AP200 VoIP Gateway Series Design Features & Concept 2002. 3.5 AddPac R&D Center Contents Design Features Design Specifications AP200 Series QoS Features AP200 Series PSTN Backup Features AP200 Series Easy

More information

IP Telephony: Review and Implementation

IP Telephony: Review and Implementation IP Telephony: Review and Implementation by Agul Kaul This project report is submitted to the Department of Electrical Engineering and Computer Science and the Faculty of the Graduate School of the University

More information

VoIP Telephone Adapter User s Manual

VoIP Telephone Adapter User s Manual VoIP Telephone Adapter User s Manual Last Update: 2008/10/10 1 Introduction...3 1.1 Product Overview (Single Phone Port Model)...3 1.2 Product Overview (Dual Phone Port Model)...4 2 IVR Interface for TA...6

More information

Cisco Unified Communications 500 Series Model 540 for Small Business

Cisco Unified Communications 500 Series Model 540 for Small Business Reference Guide Cisco Unified Communications 500 Series Model 540 for Small Business Reference Guide September, 2011 For further information, questions and comments please contact ccbu-pricing@cisco.com

More information

Adapter GL386. User Manual is available in other languages at

Adapter GL386. User Manual is available in other languages at Adapter GL386 User Manual is available in other languages at www.glipfone.com GL386 User Manual Contents: Chapter 1 Introduction ---------------------------------------------------------------- 1 Chapter

More information

IMPLEMENTING CISCO VOICE COMMUNICATIONS AND QOS Volume 1

IMPLEMENTING CISCO VOICE COMMUNICATIONS AND QOS Volume 1 IMPLEMENTING CISCO VOICE COMMUNICATIONS AND QOS Volume 1 Course Introduction Overview Learner Skills and Knowledge Course Goal and Course Flow Additional References Cisco Glossary of Terms Your Training

More information

ehealth and VoIP Overview

ehealth and VoIP Overview ehealth and VoIP Overview Voice over IP (VoIP) configurations can be very complex. Your network could contain a variety of devices, applications, and configuration capabilities to support voice traffic.

More information

for SS7 VoIP Gateways

for SS7 VoIP Gateways Smart Web Manager for SS7 VoIP Gateways www.addpac.com AddPac Technology 2013, Sales and Marketing Contents SS7 VoIP Gateway Service Diagram SS7 VoIP Gateway Main Features SS7 VoIP Gateway Features SS7

More information

SIP TRUNK HARDWARE, INSTALLATION AND MAINTENANCE MANUAL

SIP TRUNK HARDWARE, INSTALLATION AND MAINTENANCE MANUAL NOTICE Note that when converting this document from its original format to a.pdf file, some minor font and format changes may occur causing slight variations. When viewing and printing this document, we

More information

IP Addressing and Subnetting. 2002, Cisco Systems, Inc. All rights reserved.

IP Addressing and Subnetting. 2002, Cisco Systems, Inc. All rights reserved. IP Addressing and Subnetting 2002, Cisco Systems, Inc. All rights reserved. 1 Objectives Upon completion, you will be able to: Discuss the Types of Network Addressing Explain the Form of an IP Address

More information

Jive Core: Platform, Infrastructure, and Installation

Jive Core: Platform, Infrastructure, and Installation Jive Core: Platform, Infrastructure, and Installation Jive Communications, Inc. 888-850-3009 www.getjive.com 1 Overview Jive hosted services are run on Jive Core, a proprietary, cloud-based platform. Jive

More information

Introduction to Routing and Packet Forwarding. Routing Protocols and Concepts Chapter 1

Introduction to Routing and Packet Forwarding. Routing Protocols and Concepts Chapter 1 Introduction to Routing and Packet Forwarding Routing Protocols and Concepts Chapter 1 1 1 Objectives Identify a router as a computer with an OS and hardware designed for the routing process. Demonstrate

More information

Linksys SPA2102 Router Configuration Guide

Linksys SPA2102 Router Configuration Guide Linksys SPA2102 Router Configuration Guide Dear 8x8 Virtual Office Customer, This Linksys guide provides instructions on how to configure the Linksys SPA2102 as a router. You only need to configure your

More information

Curso de Telefonía IP para el MTC. Sesión 4-1 Tipos de llamadas. Mg. Antonio Ocampo Zúñiga

Curso de Telefonía IP para el MTC. Sesión 4-1 Tipos de llamadas. Mg. Antonio Ocampo Zúñiga Curso de Telefonía IP para el MTC Sesión 4-1 Tipos de llamadas Mg. Antonio Ocampo Zúñiga Call Types Local: Does not traverse the WAN or PSTN. On-net: Occurs between two telephones on the same data network.

More information

Convergence Technologies Professional (CTP) Course 1: Data Networking

Convergence Technologies Professional (CTP) Course 1: Data Networking Convergence Technologies Professional (CTP) Course 1: Data Networking The Data Networking course teaches you the fundamentals of networking. Through hands-on training, you will learn the vendor-independent

More information

Configuration Notes 290

Configuration Notes 290 Configuring Mediatrix 41xx FXS Gateway with the Asterisk IP PBX System June 22, 2011 Proprietary 2011 Media5 Corporation Table of Contents Introduction... 3 About Mediatrix 41xx Series FXS Gateways...

More information

In addition to our VoiceDirector hardware products, the following SIP broadband devices are also compatible with VoiceDirector:

In addition to our VoiceDirector hardware products, the following SIP broadband devices are also compatible with VoiceDirector: Device Compatibility Along with the full range of VoiceDirector devices we offer, a number of other SIP telephony products are compatible with the VoiceDirector corporate calling solution In addition to

More information

Configuration Notes 0217

Configuration Notes 0217 PBX Remote Line Extension using Mediatrix 1104 and 1204 Introduction... 2 Application Scenario... 2 Running the Unit Manager Network (UMN) Software... 3 Configuring the Mediatrix 1104... 6 Configuring

More information

Avaya IP Office 8.1 Configuration Guide

Avaya IP Office 8.1 Configuration Guide Avaya IP Office 8.1 Configuration Guide Performed By tekvizion PVS, Inc. Contact: 214-242-5900 www.tekvizion.com Revision: 1.1 Date: 10/14/2013 Copyright 2013 by tekvizion PVS, Inc. All Rights Reserved.

More information

640-461: Introducing Cisco Voice and Unified Communications Administration (ICOMM) v8.0 Course Introduction

640-461: Introducing Cisco Voice and Unified Communications Administration (ICOMM) v8.0 Course Introduction 640-461: Introducing Cisco Voice and Unified Communications Administration (ICOMM) v8.0 Course Introduction Course Introduction Module 01 - Overview of Cisco Unified Communications Solutions Understanding

More information

Cisco Unified Communications 500 Series

Cisco Unified Communications 500 Series Cisco Unified Communications 500 Series IP PBX Provisioning Guide Version 1.0 Last Update: 02/14/2011 Page 1 DISCLAIMER The attached document is provided as a basic guideline for setup and configuration

More information

Peer-to-Peer SIP Mode with FXS and FXO Gateways

Peer-to-Peer SIP Mode with FXS and FXO Gateways Peer-to-Peer SIP Mode with FXS and FXO Gateways New Rock s SIP based VoIP gateways with FXS and FXO ports support peer-to-peer mode which has many applications in deploying enterprise multi-site telephone

More information

SIP Trunking using Optimum Business SIP Trunk Adaptor and the Cisco Call Manager Express Version 8.5

SIP Trunking using Optimum Business SIP Trunk Adaptor and the Cisco Call Manager Express Version 8.5 CISCO SIP Trunking using Optimum Business SIP Trunk Adaptor and the Cisco Call Manager Express Version 8.5 Goal The purpose of this configuration guide is to describe the steps needed to configure the

More information

Linksys Gateway SPA2100-SU Manual

Linksys Gateway SPA2100-SU Manual Linksys Gateway SPA2100-SU Manual Manuel de l'utilisateur Table of Contents Looking for Basic Setup Instructions?... 3 Most Recent Version of this Manual... 3 Advanced Setup Instructions... 4 Wiring Your

More information

Thank you for purchasing a Panasonic Pure IP-PBX. Please read this manual carefully before using this product and save this manual for future use.

Thank you for purchasing a Panasonic Pure IP-PBX. Please read this manual carefully before using this product and save this manual for future use. IP Networking Guide Model No. Pure IP-PBX KX-NCP500 KX-NCP1000 Thank you for purchasing a Panasonic Pure IP-PBX. Please read this manual carefully before using this product and save this manual for future

More information

Linksys Voice over IP Products Guide: SIP CPE for Massive Scale Deployment

Linksys Voice over IP Products Guide: SIP CPE for Massive Scale Deployment Linksys Voice over IP Products Guide: SIP CPE for Massive Scale Deployment Corporate Headquarters Linksys 121 Theory Drive Irvine, CA 92617 USA http://www.linksys.com Tel: 949 823-1200 800 546-5797) Fax:

More information

Cisco Small Business Unified Communications 300 Series

Cisco Small Business Unified Communications 300 Series Feature Reference Guide Cisco Small Business Unified Communications 300 Series Feature Reference Guide February 2012 2011 Cisco and/or its affiliates. All rights reserved. This document is Cisco Public.

More information

ZyXEL V100 Support Notes. ZyXEL V100. (V100 Softphone 1 Runtime License) Support Notes

ZyXEL V100 Support Notes. ZyXEL V100. (V100 Softphone 1 Runtime License) Support Notes ZyXEL V100 (V100 Softphone 1 Runtime License) Support Notes Version 1.00 April 2009 1 Contents Overview 1. Overview of V100 Softphone...3 2. Setting up the V100 Softphone.....4 3. V100 Basic Phone Usage.....7

More information

CRA 210 Analog Telephone Adapter 3 Ethernet Port + 2 VoIP Line + 1 PSTN Line

CRA 210 Analog Telephone Adapter 3 Ethernet Port + 2 VoIP Line + 1 PSTN Line CRA 210 Analog Telephone Adapter 3 Ethernet Port + 2 VoIP Line + 1 PSTN Line Getting Started Guide Page: 1 of 30 Table of Contents 1. WELCOME - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -

More information

4. H.323 Components. VOIP, Version 1.6e T.O.P. BusinessInteractive GmbH Page 1 of 19

4. H.323 Components. VOIP, Version 1.6e T.O.P. BusinessInteractive GmbH Page 1 of 19 4. H.323 Components VOIP, Version 1.6e T.O.P. BusinessInteractive GmbH Page 1 of 19 4.1 H.323 Terminals (1/2)...3 4.1 H.323 Terminals (2/2)...4 4.1.1 The software IP phone (1/2)...5 4.1.1 The software

More information