Expert Reference Series of White Papers. Cisco Unified Communication Manager Digit Manipulation

Size: px
Start display at page:

Download "Expert Reference Series of White Papers. Cisco Unified Communication Manager Digit Manipulation"

Transcription

1 Expert Reference Series of White Papers Cisco Unified Communication Manager Digit Manipulation COURSES

2 Cisco Unified Communication Manager Digit Manipulation Brian Mahler, CCNA, CCNA-Voice, CCNP-Voice, Contact Center Specialist, Tandberg System Engineer Introduction What is digit manipulation? When using a telephone, we address everything with phone numbers. We have phone numbers for the home, cell, work, etc. When we want to reach someone, we dial their phone number, also called the called-party number or dialed number information system (DNIS). The recipient can recognize who is calling by looking at the calling-party or automatic number identifier (ANI). Both the DNIS and the ANI can be in a wide variety of formats and need to be modified to accommodate many different networks. For example, when I call the pizza shop to order a pizza, I dial a 10-digit number, the called-party number or DNIS, (801) , and when they answer my call, they will see my calling-party or ANI, (801) , identifying me. These 10-digit numbers match the format of the North American dial-plan. Inside a corporation, we typically have a private dial-plan that is unique to our business. For example, our office might use three-, four-, or five-digit extensions, (we will use four-digit extensions in this paper), thus the calledparty number we dial to reach a fellow worker would be only four digits long, and our calling-party number would also be four digits long. However, if I wanted to call the pizza shop, a four-digit number would not work; I have to dial that 10-digit number to go across the public telephone switch network (PSTN) and my callingparty number would have to be changed from a four-digit internal number to a 10-digit number identifying me to the pizza shop. This is the very purpose of digit manipulation, to modify either the called-party number or the calling-party number so calls can be routed across a variety of telephone networks. This process of manipulating the calling-party or called-party numbers is called digit manipulation, and this paper will explain the how and where this can be done within Cisco Unified Communication Manager (UCM). Why do we need it? We need digit manipulation any time our calls will traverse multiple dial-plans. In North America, we have the North American Dial Plan, also known as the public switch telephone network (PSTN), and most countries have their own dial plans as well, and every corporation with a phone system has its own private dial-plan within its network. It is very common for a single phone call to traverse multiple dial-plans. For example, when Company A in North America, calls Company B in Germany, the call will traverse four separate dial-plans as follows: Company A Private dial-plan > North American Dial-Plan > Germany Dial-Plan > Company B private Dial-Plan. Each one of these separate dial-plans might need to modify the called-party or calling- Copyright 2013 Global Knowledge Training LLC. All rights reserved. 2

3 party number so the call can be processed, forwarding the call on to the next hop and, eventually, to the party you want to talk with. Where can digit manipulation be done? The digit manipulation on our corporate private dial-plan can be done at the edge of our phone network or in the core. In this paper, we will focus on digit manipulation done in the core of our telephone network, using the UCM as our core phone system. UCM Digit Manipulation Components I will first explain the individual components that perform the digit manipulation, then I will use those components to explain the call flow from two different perspectives, inbound, and outbound calls. Digit manipulation in UCM can occur in multiple locations. Figures 1 and 2 below illustrate the flow and locations that digit manipulation can possibly occur for both outbound and inbound calls. Realistically, it should never be done at each point; that would be a nightmare to troubleshoot. Remember these are the different places digit manipulation can occur. I recommend that you keep it simple and only perform digit manipulation in one, or at most, two locations. IP Phone Translation Pattern (Optional) Route Pattern Route List/ Route Group Device Figure 1. Outbound Call Flow Digit Manipulation Locations Trunk Gateway Digit Analysis Translation Pattern Phone Figure 2. Inbound Call Flow Digit Manipulation Locations Digit Manipulation Tools Digit manipulation is performed using various tools built into UCM, and each of these tools is available in many different locations as well. Let s go over each of the digit manipulation tools available. At several locations within the call flow, the following three steps can be performed on either the calling-party number or the called-party number, or both. An administrator can choose to perform a single step or any combination of the three. The thing to remember is that they will always be performed in the same order, and the results of the first step are sent to the second step for processing, and the result of the second step are sent to the third step for processing. Each step is optional, but the order of the steps is fixed. Copyright 2013 Global Knowledge Training LLC. All rights reserved. 3

4 Discard Digits (Step 1) Discard Digit Instruction (DDI) does exactly that: discards or throws away digits on the front end (left side) of the number. This is the first of three steps. An administrator can choose to discarded from the pull-down menu; he or she can choose to discard an access code, country codes, dialing, or any one of 30+ possibilities. For example, in a route pattern such as , if I select the Pre-Dot option, the DDI would throw away whatever was in front of the dot (the nine). This simple discard would remove the access code 9 prior to forwarding it on to step 2. Transformation Mask (Step 2) Transformation mask is the second digit manipulation step, which can modify either the calling-party number or called-party number of a call. Transformations that modify the calling number are calling-party transformations, and transformations that modify a called number are called called-party number transformations. They both work the same, so let s detail one to see how they work. The calling party transformation allows you to manipulate the calling-party number for outgoing calls. When configuring calling-party or called-party transformations, a transformation mask operation allows the suppression of leading digits, the change of some digits while leaving others unmodified, and the insertion of leading digits. A transformation mask requires two pieces of information: the number that you want to mask and the mask itself. In the transformation mask operation, UCM aligns the number with the mask so that the last character of the mask aligns with the last digit of the number. UCM uses the corresponding digit of the original number whenever the mask contains an X. If the original number is longer than the mask, the mask removes the leading digits. If the number is shorter than the mask, the mask prefixes any digits in front of the number. Table 1 demonstrates this. Original Number 6723 First Transformation Mask XXXX Result Second Transformation Mask 203XXX0000 Result Third Transformation Mask XXX1234 Result Table 1. Transformation Mask Operation If the first transformation starts with the number 6723 and applies a transformation mask of XXXX, the end result would be Since the mask is longer than the original number, it will prefix the digits in front of the original number. Copyright 2013 Global Knowledge Training LLC. All rights reserved. 4

5 The second transformation starts with and applied the mask of 203xxx0000, which produces a result of The mask replaces the entire original number except for the three fives due to the Xs. The third transformation begins with the number and applies the mask of xxx1234, which produces a result of Since the mask is shorter than the original number, the extra digits on the front are discarded, keeping the next original three digits due to the Xs, and replacing the original last four digits with Prefixing digits (Step 3) The third and final step of digit manipulation is prefixing digits, which adds digits in front of the original number. If, for example, I dial a seven-digit number ( ), I could add a prefix of 808 (the area code), and the final result of prefixing would be , which would then be forwarded to its next destination. Figure 3 shows how each of the three-digit manipulation steps is applied to the called-party number on the web page. These same settings are available for the calling party number as well. Figure 3. How the Three-Digit MNanipulation Steps Are Applied to the Called-Party Number Calling- and Called-Party Transformation CSS Calling-party and called-party transformations CSS use the same three-digit manipulation steps (discarding, transformation mask, and prefixing) described above. The only difference is that you define how the manipulation will take place on a per route pattern level, meaning route pattern A gets digit manipulation process one while route pattern B gets digit manipulation two. A CSS contains one or more partitions, and each partition contains one or more route patterns; thus, a transformation CSS is a group of several route patterns, each with its own unique digit manipulation. This gives us flexibility when performing digit manipulation on either the calling-party or called-party number. These are all the core digit manipulation tools available at our disposal. Let s put them into use in both the outbound and inbound calls. Outbound Call Flow Originating device (i.e., phones) The originating device has limited digit manipulation capabilities; it can only manipulate the calling-party number on external calls, which it does by applying an external phone number mask to the internal calling-party number. This mask works similarly to the transformation mask. For example, if my internal calling-party number Copyright 2013 Global Knowledge Training LLC. All rights reserved. 5

6 is 4001, and I have an external phone number mask of XXXX, then the resulting calling-party number sent out on an external call is If the mask field is left blank, then the original calling-party number will be sent (even if it is only 4 digits long). This setting is configured at the line settings (Figure 4) and can be unique for each line on a phone. Figure 4. Applying an External Phone Number Mask Translation Pattern Translations patterns (not to be confused with transformations) are unique in UCM, because after the digit manipulation is performed, the called-party number is then sent back through the digit analysis process again to find the best match and its final destination. This is the only component in UCM that goes back through the digit analysis process after the translation is done, and it can be a very powerful tool. In Figure 5, you can see that both the calling-party and the called-party number can be modified independently via the same three-step digit manipulation operation explained above (discarding, transformation, and prefixing). Copyright 2013 Global Knowledge Training LLC. All rights reserved. 6

7 Figure 5. Modifying the Calling-Party and Called-Party Numbers Using Translation Patterns Dial Plan Manipulation Route Pattern The route pattern can modify both the calling-party and the called-party numbers, and the manipulation defined here applies to all dialed numbers that match the route pattern. For the calling-party number, we can choose to use the calling-party external phone number mask, which was defined on the originating phone line settings, or we can choose to use the three-step digit manipulation process defined here. Copyright 2013 Global Knowledge Training LLC. All rights reserved. 7

8 Figure 6. Modifying the Calling-Party and Called-Party Numbers Using Route Patterns Route List/Group The route list also can modify both the calling-party and the called-party numbers, and the manipulation defined here applies equally to all calls sent through this route list. We have the same three-step digit manipulation operation for both the calling-party and the called-party number explained previously. Note: If a digit manipulation is defined at the route list level, then the digit manipulation defined at the route pattern level is ignored. Copyright 2013 Global Knowledge Training LLC. All rights reserved. 8

9 Figure 7. Modifying the Calling-Party and Called-Party Numbers Using Route Lists Device Devices don t perform digit manipulation the same way that the route patterns and route list do. Devices use calling-party transformations CSS and called-party transformations CSS, which give us the ability to create custom digit manipulations on a route pattern basis, vs. defining a digit manipulation that applies to all numbers equally. We could choose to use the calling-party or called-party transformations CSS defined in the device pool instead of the one defined on the device itself (see Figure 8). You can simplify administration and save a lot of time by defining this at the device pool. It would apply to many devices and eliminate the need to define it at each individual device. Note: If a digit manipulation is defined at the device level, then the digit manipulation defined at the route list and route pattern levels are ignored. Copyright 2013 Global Knowledge Training LLC. All rights reserved. 9

10 Figure 8. Performing Digital Manipulation at the Device Level Inbound Call Flow Inbound calls have fewer locations where digit manipulation can occur than outbound calls. Trunk/Gateway Inbound calls coming into the trunk or gateway have different digit manipulation capabilities than outbound calls. There are some additional features that can simplify how the calling-party and called-party numbers are presented; these include the significant digits and prefix DN settings. Significant Digits (Step 1) The significant digits parameter informs the UCM of the number of digits of the called-party number to keep, meaning if the PSTN is providing 10 digits (e.g., ) inbound, and the UCM significant digits is set to four, then UCM keeps only the last four digits (1234), discarding the rest before forwarding it on to the next step, which is prefix DN. Note: This is only applicable to the inbound called-party number. Prefix DN (Step 2) Prefix DN is the same as prefix digits used in the transformations. The number in the prefix field is added to the front of the called-party number that was a result of the significant digits. Note: This is only applicable to the inbound called-party number. Incoming Calling-Party Settings Incoming calling-party settings uses the call type (subscriber, national, international, or unknown) to dictate what is striped and what is prefixed on the inbound calling-party number. This is the first location where an administrator can modify the calling-party number on an inbound call. Copyright 2013 Global Knowledge Training LLC. All rights reserved. 10

11 Incoming Called-Party Settings Incoming called-party settings uses the call type (Subscriber, National, International, or Unknown) to dictate what will be striped and what will be prefixed on the inbound called-party number. This may seem redundant to the significant digits and prefixing, but since it is based on the call type, it gives a little more flexibility and granular control. This concludes the device-level digit manipulations, and the call is forwarded to the digit analysis process for call routing. Translation Pattern If the digit analysis process matches a translations pattern, the translation pattern will perform the same threestep digit manipulation previously described. After the digit manipulation operations are completed for both the called-party and calling-party, the translation pattern sends the call back to the digit analysis process for best match analysis and call routing. This ability to send the call back to the digit analysis process makes translation patterns extremely powerful. Figure 9 shows that both the calling-party and the called-party number can be modified independently via the three-step digit manipulation explained previously (discarding, transformation, and prefixing). Copyright 2013 Global Knowledge Training LLC. All rights reserved. 11

12 Figure 9. Modifying the Calling-Party and Called-Party Numbers Using Translation Patterns Phone Digit manipulation with inbound calls to the phone is limited to manipulating the calling-party digits that are displayed on the phones screen. The only tool available for this is the calling-party transformation CSS located at the device settings (Figure 10). Using transformation CSS, we get a lot of flexibility to modify the calling-party number to a per calling-party basis. I don t ever see a need to go to this extreme, but we do have the capability. Copyright 2013 Global Knowledge Training LLC. All rights reserved. 12

13 Figure 10. Modifying Inbound Call Display Using the Calling-Party Transformation CSS Summary We have reviewed why we would need to modify either the calling-party or called-party number, or both, to accommodate the various dial-plans that a phone call might encounter. We also reviewed the tools available to us as well as the various locations the modifications can be performed, from within UCM. Cisco gives us the flexibility to perform the digit manipulation to meet the requirements of any network. Learn More To learn more about how you can improve productivity, enhance efficiency, and sharpen your competitive edge, Global Knowledge suggests the following courses: CIPT1 - Implementing Cisco Unified Communications IP Telephony Part 1 v8.0 CIPT2 - Implementing Cisco Unified Communications IP Telephony Part 2 v8.0 ACUCW1 - Administering Cisco Unified Communications Workspace Part 1: Basic v8.0 ACUCW2 - Administering Cisco Unified Communications Workspace Part 2: Advanced v8.0 Visit or call COURSES ( ) to speak with a Global Knowledge training advisor. About the Author Brian Mahler, CCNA, CCNA-Voice, CCNP-Voice, Contact Center Specialist, Tandberg System Engineer, operates a consulting company called NICS, LLC for over ten years, offering consulting services specializing in the area of Voice and Video over IP, and Quality of Service. Copyright 2013 Global Knowledge Training LLC. All rights reserved. 13

Expert Reference Series of White Papers. Cisco TelePresence Zones

Expert Reference Series of White Papers. Cisco TelePresence Zones Expert Reference Series of White Papers Cisco TelePresence Zones 1-800-COURSES www.globalknowledge.com Cisco TelePresence Zones Brian R Mahler, CCNA, CCNP, CCNA-Voice, CCNP-Voice, Tandberg Systems Engineer

More information

URI Dialing. Set Up URI Dialing

URI Dialing. Set Up URI Dialing Cisco Unified Communications Manager supports dialing using directory URIs for call addressing. Directory URIs look like email addresses and follow the username@host format where the host portion is an

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

Call Setup and Digit Manipulation

Call Setup and Digit Manipulation Call Setup and Digit Manipulation End-to-End Calls This topic explains how routers interpret call legs to establish end-to-end calls. End-to-End Calls IP Telephony 2005 Cisco Systems, Inc. All rights reserved.

More information

Expert Reference Series of White Papers. Binary and IP Address Basics of Subnetting

Expert Reference Series of White Papers. Binary and IP Address Basics of Subnetting Expert Reference Series of White Papers Binary and IP Address Basics of Subnetting 1-800-COURSES www.globalknowledge.com Binary and IP Address Basics of Subnetting Alan Thomas, CCNA, CCSI, Global Knowledge

More information

TECH ARTICLE Date: 03/04/08

TECH ARTICLE Date: 03/04/08 TechnicalArticle.doc Version1.0 23/10/07 TECH ARTICLE Date: 03/04/08 Topic / Issue: Written By: Configuring ISDN Settings on Quadro PBX Chris Pulsford This document describes the configuration of the ISDN

More information

Expert Reference Series of White Papers. The Basics of Configuring and Using Cisco Network Address Translation

Expert Reference Series of White Papers. The Basics of Configuring and Using Cisco Network Address Translation Expert Reference Series of White Papers The Basics of Configuring and Using Cisco Network Address Translation 1-800-COURSES www.globalknowledge.com The Basics of Configuring and Using Cisco Network Address

More information

Dial Peer. Example: Dial-Peer Configuration

Dial Peer. Example: Dial-Peer Configuration Configuring Dial Peers Understanding Dial Peers This topic describes dial peers and their applications. Understanding Dial Peers A dial peer is an addressable call endpoint. Dial peers establish logical

More information

Administering Cisco Unified Communications Manager (ACUCM) v10.0

Administering Cisco Unified Communications Manager (ACUCM) v10.0 Administering Cisco Unified Communications Manager (ACUCM) v10.0 This five day instructor-led course presented by Cisco training partners to their end customers. Students will learn the basic procedures

More information

Communications Manager Administration **Developed by Skyline ATS** Communications Manager Administration**Developed by Skyline ATS**

Communications Manager Administration **Developed by Skyline ATS** Communications Manager Administration**Developed by Skyline ATS** Course: Communications Manager Administration Duration: 3 Day Hands-On Lab & Lecture Course Price: $ 2,695.00 Learning Credits: 27 Description: Communications Manager Administration for Version 9.0 (CMA90)

More information

Hotline. Configure Hotline

Hotline. Configure Hotline This chapter provides information about the hotline feature which extends the Private Line Automatic Ringdown (PLAR) feature, which allows you to configure a phone so that when the user goes off hook (or

More information

Acano solution. Third Party Call Control Guide. March 2015 76-1055-01-E

Acano solution. Third Party Call Control Guide. March 2015 76-1055-01-E Acano solution Third Party Call Control Guide March 2015 76-1055-01-E Contents Contents 1 Introduction... 3 1.1 How to Use this Guide... 3 1.1.1 Commands... 4 2 Example of Configuring a SIP Trunk to CUCM...

More information

Implementing Cisco Unified Communications Manager Part 1, Volume 1

Implementing Cisco Unified Communications Manager Part 1, Volume 1 Implementing Cisco Unified Communications Manager Part 1, Volume 1 Course Introduction Overview Learner Skills and Knowledge Course Goal and Course Flow Additional Cisco Glossary of Terms Your Training

More information

Administering Cisco Unified Communication Manager v8.0 (ACUCM v8.0) Training

Administering Cisco Unified Communication Manager v8.0 (ACUCM v8.0) Training Administering Cisco Unified Communication Manager v8.0 (ACUCM v8.0) Training Module 1: Introduction to IP Telephony Lesson 1: Exploring IP Telephony Traditional Voice versus IP Telephony Clustering Overview

More information

Building a Scalable Numbering Plan

Building a Scalable Numbering Plan Building a Scalable Numbering Plan Scalable Numbering Plan This topic describes the need for a scalable numbering plan in a VoIP network. Dial Plans Dial plans contain specific dialing patterns for a user

More information

Cisco Unified Communications IP Telephony Part 1 (CIPT1) v8.0. Course Objectives. Associated Certification. Required Exam(s) Price.

Cisco Unified Communications IP Telephony Part 1 (CIPT1) v8.0. Course Objectives. Associated Certification. Required Exam(s) Price. Cisco Unified Communications IP Telephony Part 1 (CIPT1) v8.0 Implementing Cisco Unified Communications Manager, Part 1 (CIPT1) v8.0 prepares you for implementing a Cisco Unified Communications Manager

More information

Integrating Skype for SIP with UC500

Integrating Skype for SIP with UC500 Integrating Skype for SIP with UC500 Version 1.1 2008 Cisco Systems, Inc. All rights reserved. 1 TABLE OF CONTENTS 1 OVERVIEW... 3 1.1 INTRODUCTION... 3 1.2 SCOPE... 3 1.3 REVISION CONTROL... 3 1.4 RESTRICTIONS...

More information

Version dated 25/11/2014. 1.Course Title. NATO Voice over IP Foundation Course. 2.Identification Number (ID) 3. Purpose of the Course

Version dated 25/11/2014. 1.Course Title. NATO Voice over IP Foundation Course. 2.Identification Number (ID) 3. Purpose of the Course 1.Course Title Version dated 25/11/2014 NATO Voice over IP Foundation Course 2.Identification Number (ID) 095 3. Purpose of the Course There are a number of new technologies (to NATO) that are encompassed

More information

Configuration Manager

Configuration Manager After you have installed Unified Intelligent Contact Management (Unified ICM) and have it running, use the to view and update the configuration information in the Unified ICM database. The configuration

More information

Enabling Users for Lync services

Enabling Users for Lync services Enabling Users for Lync services 1) Login to collaborate.widevoice Server as admin user 2) Open Lync Server control Panel as Run As Administrator 3) Click on Users option and click Enable Users option

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

Cisco Expressway Basic Configuration

Cisco Expressway Basic Configuration Cisco Expressway Basic Configuration Deployment Guide Cisco Expressway X8.1 D15060.03 August 2014 Contents Introduction 4 Example network deployment 5 Network elements 6 Internal network elements 6 DMZ

More information

Implementing Cisco Unified Communications Manager, Part 1 (CIPT1) Foundation Learning Guide Second Edition

Implementing Cisco Unified Communications Manager, Part 1 (CIPT1) Foundation Learning Guide Second Edition Implementing Cisco Unified Communications Manager, Part 1 (CIPT1) Foundation Learning Guide Second Edition Josh Finke Dennis Hartmann Cisco Press 800 East 96th Street Indianapolis, IN 46240 ii Implementing

More information

SUBNETTING SCENARIO S

SUBNETTING SCENARIO S SUBNETTING SCENARIO S This white paper provides several in-depth scenario s dealing with a very confusing topic, subnetting. Many networking engineers need extra practice to completely understand the intricacies

More information

Configuration Checklist for Immediate Divert

Configuration Checklist for Immediate Divert CHAPTER 27 The (idivert) feature allows you to immediately divert a call to a voice-messaging system. When the call gets diverted, the line becomes available to make or receive new calls. This chapter

More information

SPAMfighter Mail Gateway

SPAMfighter Mail Gateway SPAMfighter Mail Gateway User Manual Copyright (c) 2009 SPAMfighter ApS Revised 2009-05-19 1 Table of contents 1. Introduction...3 2. Basic idea...4 2.1 Detect-and-remove...4 2.2 Power-through-simplicity...4

More information

Configuring Avaya BCM 6.0 for Spitfire SIP Trunks

Configuring Avaya BCM 6.0 for Spitfire SIP Trunks Configuring Avaya BCM 6.0 for Spitfire SIP Trunks This document is a guideline for configuring Spitfire SIP trunks onto Business Communication Manager release 6.0 and includes the settings required for

More information

Application Notes for Configuring Microsoft Office Communications Server 2007 R2 and Avaya IP Office PSTN Call Routing - Issue 1.0

Application Notes for Configuring Microsoft Office Communications Server 2007 R2 and Avaya IP Office PSTN Call Routing - Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Configuring Microsoft Office Communications Server 2007 R2 and Avaya IP Office PSTN Call Routing - Issue 1.0 Abstract These Application

More information

CISCO UNIFIED COMMUNICATIONS MANAGER

CISCO UNIFIED COMMUNICATIONS MANAGER CISCO UNIFIED COMMUNICATIONS MANAGER V10 UPDATED TOPICS LOCAL ROUTE GROUP ENHANCEMENT Multiple Local Route Groups can be associated with Route Groups for Emergency Dialing. In releases 8 and 9, administrators

More information

ShoreTel & AMTELCO Infinity Console via SIP Trunking (Native)

ShoreTel & AMTELCO Infinity Console via SIP Trunking (Native) Product: ShoreTel AMTELCO Infinity Console I n n o v a t i o n N e t w o r k A p p N o t e IN-15063 Date : October, 2015 System version: ShoreTel 14.2 ShoreTel & AMTELCO Infinity Console via SIP Trunking

More information

ADMINISTERING CISCO UNIFIED COMMUNICATION MANAGER

ADMINISTERING CISCO UNIFIED COMMUNICATION MANAGER ADMINISTERING CISCO UNIFIED COMMUNICATION MANAGER AND UNITY CONNECTION V8.0 (ACUCM+AUC) COURSE OVERVIEW: Administering Cisco Unified Communications Manager (ACUCM) provides system administrators and networking

More information

iseries TCP/IP routing and workload balancing

iseries TCP/IP routing and workload balancing iseries TCP/IP routing and workload balancing iseries TCP/IP routing and workload balancing Copyright International Business Machines Corporation 2000, 2001. All rights reserved. US Government Users Restricted

More information

Configuring the PBX Call Routing Table for outbound calls (with security against unsecured calls)

Configuring the PBX Call Routing Table for outbound calls (with security against unsecured calls) Configuring the PBX Call Routing Table for outbound calls (with security against unsecured calls) The Quadro s Call Routing Table (CRT) defines how incoming and outgoing calls will be handled by the Quadro.

More information

Implementing Cisco IP Telephony & Video, Part 2 CIPTV2 v1.0; 5 Days; Instructor-led

Implementing Cisco IP Telephony & Video, Part 2 CIPTV2 v1.0; 5 Days; Instructor-led Implementing Cisco IP Telephony & Video, Part 2 CIPTV2 v1.0; 5 Days; Instructor-led Course Description Implementing Cisco IP Telephony & Video, Part 2 (CIPTV2) v1.0 is a five-day course that prepares the

More information

ESI SIP Trunking Installation Guide

ESI SIP Trunking Installation Guide ESI SIP Trunking Installation Guide 0450-1227 Rev. B Copyright 2009 ESI (Estech Systems, Inc.). Information contained herein is subject to change without notice. ESI products are protected by various U.S.

More information

Internet Telephony PBX System

Internet Telephony PBX System Internet Telephony PBX System T1/E1 Gateway With IP PBX Application Copyright PLANET Technology Corporation. All rights reserved. Case 35: With IP PBX Application Head Office E1 PABX interconnect with

More information

Reduce Mobile Phone Expense with Avaya Unified Communications

Reduce Mobile Phone Expense with Avaya Unified Communications Reduce Mobile Phone Expense with Avaya Unified Communications Table of Contents Section 1: Reduce Inbound Minutes... 2 Section 2: Reduce Outbound Minutes... 3 Section 3: Take Greater Advantage of Free

More information

Special-Purpose Connections

Special-Purpose Connections Special-Purpose Connections Connection Commands This topic identifies different special-purpose connection commands. Special-Purpose Connection Commands connection plar Associates a voice port directly

More information

nexvortex Setup Template

nexvortex Setup Template nexvortex Setup Template ZULTYS, INC. April 2013 5 1 0 S P R I N G S T R E E T H E R N D O N V A 2 0 1 7 0 + 1 8 5 5. 6 3 9. 8 8 8 8 Introduction This document is intended only for nexvortex customers

More information

Sametime Unified Telephony Lite Client:

Sametime Unified Telephony Lite Client: Sametime Version 8.5.2 From Zero to Hero Sametime Unified Telephony Lite Client: Configuring SIP trunks to third-party audio/video equipment Contents Edition Notice...4 1 Introduction...5 1.1 What is Sametime

More information

Communications Transformations 2: Steps to Integrate SIP Trunk into the Enterprise

Communications Transformations 2: Steps to Integrate SIP Trunk into the Enterprise Communications Transformations 2: Steps to Integrate SIP Trunk into the Enterprise The Changing Landscape IP-based unified communications is widely deployed in enterprise networks, both for internal calling

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

Abstract. Avaya Solution & Interoperability Test Lab

Abstract. Avaya Solution & Interoperability Test Lab Avaya Solution & Interoperability Test Lab Application Notes for Configuring SIP Trunking between the Verizon Business VoIP Service with IP Trunking and Avaya Communication Manager Branch Edition Issue

More information

Application Notes for Configuring Avaya IP Office 8.1 with Colt VoIP Access service Issue 1.0

Application Notes for Configuring Avaya IP Office 8.1 with Colt VoIP Access service Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Configuring Avaya IP Office 8.1 with Colt VoIP Access service Issue 1.0 Abstract These Application Notes describe the procedures for configuring

More information

Expert Reference Series of White Papers. Basics of IP Address Subnetting

Expert Reference Series of White Papers. Basics of IP Address Subnetting Expert Reference Series of White Papers Basics of IP Address Subnetting 1-800-COURSES www.globalknowledge.com Basics of IP Address Subnetting Norbert Gregorio, Global Knowledge Instructor Introduction

More information

IP TELEPHONY & UNIFIED COMMUNICATIONS

IP TELEPHONY & UNIFIED COMMUNICATIONS IP TELEPHONY & UNIFIED COMMUNICATIONS English GET IN TOUCH innovaphone AG Böblinger Str. 76 D-71065 Sindelfingen Tel. +49 7031 73009-0 Fax +49 7031 73009-9 info@innovaphone.com www.innovaphone.com YOUR

More information

Device SIP Trunking Administrator Manual

Device SIP Trunking Administrator Manual Table of Contents Device SIP Trunking Administrator Manual Version 20090401 Table of Contents... 1 Your SIP Trunking Service... 2 Terminology and Definitions... 2 PBX, IP-PBX or Key System... 2 Multi-port

More information

Configuration guide on common features of OM20 with NeeHau APP

Configuration guide on common features of OM20 with NeeHau APP Configuration guide on common features of OM20 with NeeHau APP Part 1: OM20 OM20 is a full-featured IP-Telephony system with 20 extensions, providing features such as auto-attendant, mobile extension,

More information

Configuring Mitel 3300 for Spitfire SIP Trunks

Configuring Mitel 3300 for Spitfire SIP Trunks Configuring Mitel 3300 for Spitfire SIP Trunks This document is a guideline for configuring Spitfire SIP trunks onto Mitel 3300 and includes the settings required for Inbound DDI routing and Outbound CLI

More information

ACCELERATOR 6.3 ASTERISK 1.4 INTEGRATION GUIDE

ACCELERATOR 6.3 ASTERISK 1.4 INTEGRATION GUIDE ACCELERATOR 6.3 ASTERISK 1.4 INTEGRATION GUIDE October 2014 Tango Networks, Inc. phone: +1 469-229-6000 3801 Parkwood Blvd, Suite 500 fax: +1 469-467-9840 Frisco, Texas 75034 USA www.tango-networks.com

More information

ScopTEL TM IP PBX Software. Back to Back SIP Trunking Configuration

ScopTEL TM IP PBX Software. Back to Back SIP Trunking Configuration Back to Back SIP Trunking Configuration Usage Cases Usage Cases Implementing DNIS: SIP TIE trunks: A private network is created to dial extensions between systems using Access Codes Tandem Dialing: PSTN

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

Configuring Quadro IP PBXs with "SIP Connect"

Configuring Quadro IP PBXs with SIP Connect Configuring Quadro IP PBXs with "SIP Connect" Revision: 1.0 Abstract: This document describes how to configure the Quadro IP PBXs to use the IP-PSTN service from SIP Connect PAGE 1 Document Revision History

More information

Configuring a Pure-IP SIP Trunk in Lync 2013

Configuring a Pure-IP SIP Trunk in Lync 2013 Configuring a Pure-IP SIP Trunk in Lync 2013 Contents Configuring a Pure-IP SIP Trunk in Lync 2013... 1 Introduction - Product version: Microsoft Lync Server 2013... 2 Pure-IP SIP Trunk configuration tasks...

More information

CIPT1_8- Implementing Cisco Unified Communications Manager Part 1

CIPT1_8- Implementing Cisco Unified Communications Manager Part 1 CIPT1_8- Implementing Cisco Unified Communications Manager Part 1 Description Implementing Cisco Unified Communications Manager, Part 1 (CIPT1) v8.0 prepares you for implementing a Cisco Unified Communications

More information

Installation and Configuration Manual

Installation and Configuration Manual Beacon Office Installation and Configuration Manual Version - 2.5(1) Radianta Inc. September 2008 Radianta, Inc. Beacon Office Page 2 Table of Contents Introduction... 4 What is Beacon Office... 4 How

More information

Automatic Routing of Inbound Faxes with Open Text Network Fax Servers

Automatic Routing of Inbound Faxes with Open Text Network Fax Servers Automatic Routing of Inbound Faxes with Open Text Network Fax Servers Functional T1, Full T1, Fractional E1, Full E1, ISDN, DID, and PBX Integration Open Text Fax and Document Distribution Group October

More information

Hosted PBX. Benefit from business-grade Unified Communications, easier network management, and enhanced employee productivity.

Hosted PBX. Benefit from business-grade Unified Communications, easier network management, and enhanced employee productivity. UNIFIED COMMUNIC ATIONS Hosted PBX Benefit from business-grade Unified Communications, easier network management, and enhanced employee productivity. Business IT departments increasingly want to reduce

More information

Cisco Unified Video Conferencing Configuration

Cisco Unified Video Conferencing Configuration Cisco Unified Video Conferencing Configuration This topic provides a reference configuration for Cisco Unified Video Conferencing within a Cisco Unified Communications deployment. The information is based

More information

Integrating Citrix EasyCall Gateway with SwyxWare

Integrating Citrix EasyCall Gateway with SwyxWare Integrating Citrix EasyCall Gateway with SwyxWare The EasyCall Gateway has been tested for interoperability with Swyx SwyxWare, versions 6.12 and 6.20. These integration tests were done by using EasyCall

More information

SIP Internet Telephony Gateway

SIP Internet Telephony Gateway SIP Internet Telephony Gateway VIP - 2 / 4 / 8 / 16 / 24 Series Peer-to-Peer Quick Configuration for VIP-450 Copyright PLANET Technology Corporation. All rights reserved. Scenarios explain: Peer-to-Peer

More information

"Charting the Course... Implementing Cisco IP Telephony & Video, Part 1 v1.0 ( CIPTV1 ) Course Summary

Charting the Course... Implementing Cisco IP Telephony & Video, Part 1 v1.0 ( CIPTV1 ) Course Summary Description Course Summary Implementing Cisco IP Telephony & Video, Part 1 (CIPTV1) v1.0 is a five-day course that prepares the learner for implementing a Cisco Collaboration solution at a single-site

More information

nexvortex SIP Trunking

nexvortex SIP Trunking nexvortex SIP Trunking January 2015 510 SPRING STREET HERNDON VA 20170 +1 855.639.8888 Copyright nexvortex 2014 This document is the exclusive property of nexvortex, Inc. and no part may be disclosed,

More information

Internet Telephony PBX System

Internet Telephony PBX System Internet Telephony PBX System GSM Gateway PSTN call busy forward to GSM Configuration Copyright PLANET Technology Corporation. All rights reserved. Case 32: PSTN call busy forward to GSM Configuration

More information

Introducing Cisco Voice and Unified Communications Administration Volume 1

Introducing Cisco Voice and Unified Communications Administration Volume 1 Introducing Cisco Voice and Unified Communications Administration Volume 1 Course Introduction Overview Learner Skills and Knowledge Course Goal and Course Flow Additional Cisco Glossary of Terms Your

More information

Avaya Solution & Interoperability Test Lab. Abstract

Avaya Solution & Interoperability Test Lab. Abstract Avaya Solution & Interoperability Test Lab Application Notes for Avaya Aura Communication Manager 6.3, Avaya Session Manager 6.3, and Avaya Session Border Controller for Enterprise 6.3, with AT&T IP Flexible

More information

Chapter 8 Advanced Configuration

Chapter 8 Advanced Configuration Chapter 8 Advanced Configuration This chapter describes how to configure the advanced features of your ProSafe 802.11g Wireless VPN Firewall FVG318. Configuring Dynamic DNS If your network has a permanently

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

Implementing Cisco Unified Communications Manager Part 1 Course IPT1 v9.0; 5 Days, Instructor-led

Implementing Cisco Unified Communications Manager Part 1 Course IPT1 v9.0; 5 Days, Instructor-led Implementing Cisco Unified Communications Manager Part 1 Course IPT1 v9.0; 5 Days, Instructor-led Course Description Implementing Cisco Unified Communications Manager, Part 1 (IPT1) v9.0 prepares you for

More information

Integrating a Mitel 3300 ICP system with a IPCM System.

Integrating a Mitel 3300 ICP system with a IPCM System. Integrating a Mitel 3300 ICP system with a IPCM System. Version: 1.0 Created by: Francisco Piedra Date: Nov 27 th 2008 Web Site: E-Mail: info@sai.es Phone: +34935906366 Chapter 1: Mitel SIP Configuration

More information

ADTRAN SBC and Cisco Unified Call Manager SIP Trunk Interoperability

ADTRAN SBC and Cisco Unified Call Manager SIP Trunk Interoperability 6AOSSG0004-42A April 2013 Interoperability Guide ADTRAN SBC and Cisco Unified Call Manager SIP Trunk Interoperability This guide describes an example configuration used in testing the interoperability

More information

Cisco EXAM - 300-075. Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2) Buy Full Product. http://www.examskey.com/300-075.

Cisco EXAM - 300-075. Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2) Buy Full Product. http://www.examskey.com/300-075. Cisco EXAM - 300-075 Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2) Buy Full Product http://www.examskey.com/300-075.html Examskey Cisco 300-075 exam demo product is here for you to test the

More information

Passit4Sure.642-447.276Q.A

Passit4Sure.642-447.276Q.A Passit4Sure.642-447.276Q.A Number: 642-447 Passing Score: 800 Time Limit: 120 min File Version: 28.8 http://www.gratisexam.com/ 642-447 Implementing Cisco Unified Communications Manager, Part 1 v8.0 (CIPT1

More information

Enable SMTP Message Notifications in Cisco Unity Connection 8.x

Enable SMTP Message Notifications in Cisco Unity Connection 8.x Enable SMTP Message Notifications in Cisco Unity Connection 8.x Document ID: 113623 Contents Introduction Prerequisites Requirements Components Used Conventions Set up SMTP Message Notifications Configurations

More information

APPLICATION NOTE: AN10440

APPLICATION NOTE: AN10440 APPLICATION NOTE: AN10440 Integrating ShoreTel with Microsoft Lync via AudioCodes IP to IP (SIP) Microsoft Lync 2010 and AudioCodes Mediant 1000 Media Gateway Contents Overview... 2 Required Components...

More information

GSN Cloud Contact Centre Voice & Telephony Datasheet

GSN Cloud Contact Centre Voice & Telephony Datasheet GSN Cloud Contact Centre Voice & Telephony Datasheet Commercial in Confidence Reference: GSN CCC - Voice & Telephony Version: 1.1 Global Speech Networks Pty Ltd Level 8, 636 St Kilda Road Melbourne, Victoria

More information

ISDN (PRI) GSM Gateway

ISDN (PRI) GSM Gateway HG-3000/3U ISDN (PRI) GSM Gateway Cost Saving Customer Premises Equipment with Carrier Grade Performance Product Description November 2009 Contents Next Page Hypermedia HG-3000/3U 3 Contents ABSTRACT...

More information

Connecting with Vonage

Connecting with Vonage Connecting with Vonage Vonage (http://www.vonage.com/) offers telephone service using the VoIP (Voice over Internet Protocol) standard SIP (Session Initiation Protocol). The service allow users making

More information

nexvortex VOIP DISASTER RECOVERY BUSINESS SOLUTION

nexvortex VOIP DISASTER RECOVERY BUSINESS SOLUTION nexvortex VOIP DISASTER RECOVERY BUSINESS SOLUTION Terry Prime Chief Technology Officer February 2007 Copyright 2007 Introduction The telephone service is a strategic component of any business or government

More information

Cisco BE6K Solutions from TA Networks

Cisco BE6K Solutions from TA Networks Cisco BE6K Solutions from TA Networks We ve been deploying Cisco UC since 2001 get quotes online at why us We ve been advising customers on, deploying and supporting Cisco UC deployments for 15 years.

More information

2.1.2.2.2 Variable length subnetting

2.1.2.2.2 Variable length subnetting 2.1.2.2.2 Variable length subnetting Variable length subnetting or variable length subnet masks (VLSM) allocated subnets within the same network can use different subnet masks. Advantage: conserves the

More information

Using Avaya one-x Communicator Release 6.1

Using Avaya one-x Communicator Release 6.1 Using Avaya one-x Communicator Release 6.1 October 2011 Table Of Contents Introduction... 1 Logging in to the server... 3 Logging out of the server... 3 Using your feature buttons... 5 Using Avaya one-x

More information

Implementing Cisco Collaboration Devices CICD v1.0; 5 Days; Instructor-led

Implementing Cisco Collaboration Devices CICD v1.0; 5 Days; Instructor-led Implementing Cisco Collaboration Devices CICD v1.0; 5 Days; Instructor-led Course Description Implementing Cisco Collaboration Devices (CICD v1.0) is an extended hours 5-day course focusing on providing

More information

Avaya one-x Mobile User Guide for iphone

Avaya one-x Mobile User Guide for iphone Avaya one-x Mobile User Guide for iphone Release 5.2 January 2010 0.3 2009 Avaya Inc. All Rights Reserved. Notice While reasonable efforts were made to ensure that the information in this document was

More information

Avaya Aura Session Manager Overview

Avaya Aura Session Manager Overview Avaya Aura Session Manager Overview 03-603323, Issue 1 Release 1.1 May 2009 2009 Avaya Inc. All Rights Reserved. Notices While reasonable efforts were made to ensure that the information in this document

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

Channel Manager. VPI 160 Camino Ruiz, Camarillo, CA 93012-6700 (Voice) 800-200-5430 805-389-5200 (Fax) 805-389-5202 www.vpi-corp.

Channel Manager. VPI 160 Camino Ruiz, Camarillo, CA 93012-6700 (Voice) 800-200-5430 805-389-5200 (Fax) 805-389-5202 www.vpi-corp. Cisco Automa utomatic tic Recording Channel Manager VPI 160 Camino Ruiz, Camarillo, CA 93012-6700 (Voice) 800-200-5430 805-389-5200 (Fax) 805-389-5202 www.vpi-corp.com Copyright 2008 Voice Print International,

More information

Cisco CallManager 4.1 SIP Trunk Configuration Guide

Cisco CallManager 4.1 SIP Trunk Configuration Guide Valcom Session Initiation Protocol (SIP) VIP devices are compatible with Cisco Unified Communications Manager systems. For versions of Communications Manager that do not support SIP endpoints (such as

More information

How to Buy a Business Phone System

How to Buy a Business Phone System How to Buy a Business Phone System An Inside Guide to What You Need to Know When Choosing a Business Phone System Digitcom Canada Rimrock Road, Toronto, Ontario, M J A T:. - or... E: sales@digitcom.ca

More information

Grandstream Networks, Inc. How to Integrate UCM6100 with Microsoft Lync Server

Grandstream Networks, Inc. How to Integrate UCM6100 with Microsoft Lync Server Grandstream Networks, Inc. How to Integrate UCM6100 with Microsoft Lync Server Index Table of Contents OVERVIEW... 3 UCM6100 CONFIGURATION... 4 STEP 1: CREATE SIP PEER TRUNK... 4 STEP 2: CONFIGURE OUTBOUND

More information

Expert Reference Series of White Papers. The Role of Session Manager in Applying Features to Calls via Sequenced Applications, Part 1

Expert Reference Series of White Papers. The Role of Session Manager in Applying Features to Calls via Sequenced Applications, Part 1 Expert Reference Series of White Papers The Role of Session Manager in Applying Features to Calls via Sequenced Applications, Part 1 1-800-COURSES www.globalknowledge.com The Role of Session Manager in

More information

Implementing Cisco IP Telephony & Video, Part 1

Implementing Cisco IP Telephony & Video, Part 1 Course Code: CI-CIPTV1 Vendor: Cisco Course Overview Duration: 5 RRP: 2,320 Implementing Cisco IP Telephony & Video, Part 1 Overview Implementing Cisco IP Telephony & Video, Part 1 (CIPTV1) v1.0 is a five-day

More information

Implementing Intercluster Lookup Service

Implementing Intercluster Lookup Service Appendix 11 Implementing Intercluster Lookup Service Overview When using the Session Initiation Protocol (SIP), it is possible to use the Uniform Resource Identifier (URI) format for addressing an end

More information

Adjusting Voice Quality

Adjusting Voice Quality Adjusting Voice Quality Electrical Characteristics This topic describes the electrical characteristics of analog voice and the factors affecting voice quality. Factors That Affect Voice Quality The following

More information

How Cisco IT Reduced Linksys Contact Center Outsourcing

How Cisco IT Reduced Linksys Contact Center Outsourcing How Cisco IT Reduced Linksys Contact Center Outsourcing Cisco Unified Intelligent Contact Management reduces outsourcing costs by improving contact center management and support. Cisco IT Case Study /

More information

Expert Reference Series of White Papers. VMware vsphere Distributed Switches

Expert Reference Series of White Papers. VMware vsphere Distributed Switches Expert Reference Series of White Papers VMware vsphere Distributed Switches info@globalknowledge.net www.globalknowledge.net VMware vsphere Distributed Switches Rebecca Fitzhugh, VCAP-DCA, VCAP-DCD, VCAP-CIA,

More information

How Your Computer Accesses the Internet through your Wi-Fi for Boats Router

How Your Computer Accesses the Internet through your Wi-Fi for Boats Router How Your Computer Accesses the Internet through your Wi-Fi for Boats Router By default, a router blocks any inbound traffic from the Internet to your computers except for replies to your outbound traffic.

More information

Introduction to VoIP for Small and Medium Sized Businesses

Introduction to VoIP for Small and Medium Sized Businesses Introduction to VoIP for Small and Medium Sized Businesses Understanding the Options and Opportunities April 17th, 2007 Notice Copyright 2009 Metaswitch Networks. All rights reserved. Other brands and

More information

CREATE A CUSTOMER... 2 SIP TRUNK ACCOUNTS...

CREATE A CUSTOMER... 2 SIP TRUNK ACCOUNTS... Contents CREATE A CUSTOMER... 2 SIP TRUNK ACCOUNTS... 3 CREATE THE MAIN SIP TRUNK ACCOUNT... 3 SETUP THE SIP TRUNK ACCOUNT... 4 EXTRA DIDS... 7 HOW TO..... 9 BILL FOR THE SIP TRUNKING SERVICE... 9 LIMIT

More information

About. IP Centrex App for ios Tablet. User Guide

About. IP Centrex App for ios Tablet. User Guide About IP Centrex App for ios Tablet User Guide December, 2015 1 2015 by Cox Communications. All rights reserved. No part of this document may be reproduced or transmitted in any form or by any means, electronic,

More information