Leveraging Asterisk to Deliver Large Scale VoIP Services in a Carrier Environment. JR Richardson
|
|
- Amanda Webb
- 8 years ago
- Views:
Transcription
1 Leveraging Asterisk to Deliver Large Scale VoIP Services in a Carrier Environment JR Richardson
2 Early VoIP Environment Telecom Act of 1996, mass competition, Telco's needed value add features and capabilities, VoIP held great potential VoIP services very immature and costly Hardware VoIP solutions expensive with few features compared to POTS Software VoIP existed but PC platform processor speeds too low for mass scale
3 What is Asterisk? Open Source Hybrid TDM and packet voice PBX and IVR platform Runs on general computing platform Analog and Digital telephony interfaces Supports SIP, H323, MGCP, SCCP and IAX2 Dial Plan scripting bound only by imagination of administrator
4 Hot Debate: Hardware or Software DSP? Digital Signal Processing has been a barrier to low cost VoIP services Hardware solutions are high cost in nature Software DSP technology has been around for years but PC performance low PC performance has increased enough to warrant investigation into large scale soft switch solutions
5 Hardware vs. Software DSP Exercise Critical planning event for VoIP deployment, calculate PSTN access, TDM to Packet Voice transcoding Large Scale 200K line side extensions Trunk ratio 7:1 28,571 DSP channels required
6 Hardware DSP Cost Lowest price Dialogic hardware, 2xT1 PCI card, $7,999 Assuming volume 50%, $4000/2xT1 card achievable Calculation comes to $83/port ($4000/48) 28,571 trunks / 24 ports per T1 = 1191 T1 s needed / 2 T1 s per card = 596 cards 596 cards x $4000 = $2,384K
7 Software DSP Cost Digium 4xT1 cards list $1500 Assuming volume 50%, $750/4xT1 card achievable (is Mark Spencer watching?) Calculation comes to $7.80/port ($750/96) 28,571 trunks / 24 ports per T1 = 1191 T1 s needed / 4 T1 s per card = 298 cards 298 cards x $750 = $223,500 1/10 th the cost of hardware DSP cards
8 Processor Loading Sending transcoding function to central CPU has trade offs, processor loading To utilize all 96 channels on 4xT1 card, the server must be robust, driving up cost Dual Zeon 2.8GHz, 1 Gig RAM, SCSI HD With strong vendor relations, can achieve server $2000 With 1-4xT1 Digium card per server, 298 are servers needed
9 Processor Loading (cont) 298 servers x $2000 = $596K With hardware DSP cards, same server could easily handle 4 cards With 4 2xT1 cards per server, 149 servers are needed 149 servers x $2000 = $298K Cost for Hardware DSP PSTN GW $2,683K, Software PSTN GW $819K
10 Dedicated Platform Cost Cisco AS5850 Good port density with 5 CT3 s (190 T1 s), 3360 channels in one fully loaded chassis Lowest cost found $175K Calculation come to $52/port ($175K/3360) 28,572 trunks require 8.5 AS5850 s Cost for this solution $1,513K
11 Dedicated DSP vs Software DSP $3,000 $2,500 $2,384 $2,682 In M illions $2,000 $1,500 $1,000 $500 $224 $298 $596 $820 $1,513 $0 Interface Card Cost Server Cost Total Cost Hardware Dedicated DSP Chip Software DSP Purpose Built
12 Rack Cost and Space 8 racks required for software PSTN GW 4 racks required for hardware DSP 3 racks for AS5850 platform Rack kits ~$1000 Space availability more critical than cost of rack hardware
13 Power Cost Software DSP, Watts per server = 89.4KW Hardware DSP, Watts per server = 44.7KW 9 Cisco AS KW per unit = 21.6KW Average cost per KW = $4,300 Software $390K, Hardware $190K, Cisco AS5850 s $130K
14 Power Cost In Thousands $450 $400 $350 $300 $250 $200 $150 $100 $50 $0 $390 $195 Software DSP, 298 Servers Dedicated DSP Chip, 149 Servers AC Power Plant $130 Purpose Built, 9 Cisco AS5850's
15 Codec Cost Codec g.711 $0, high bandwidth 64KB Codec g.729 $10, low bandwidth 8KB Using g.729 codec in this exercise drives up cost of software DSP solution by adding $285,720 Ultimately the network architecture will determine codec usage Also codec usage can change as requirements dictate
16 Cost Includes Interface Cards, Servers, Racks and Power $3,500 $3,000 $2,881 $2,500 In M illio n s $2,000 $1,500 $1,000 $1,218 $1,503 $1,643 $500 $0 Software DSP Solution Software DSP Solution with g.729 codec Dedicated DSP Soulution Cisco AS5850 Solution VoIP Service for 28,572 PSTN Trunks
17 Exercise Conclusion Cost exercise intended to show potential cost differential between two methods of converting analog voice to packet voice (Voice quality note) When implemented properly, there is no discernable difference between the two methods
18 Architecture and Integration Large deployments scale better when hardware platforms are separated by functionality Voice Mail PSTN Gateway Applications Call Conference Customer Aggregation Dial Plan Routing
19 PSTN Gateway Asterisk can be configured to perform PSTN GW functions to create a virtual VoIP channel per T1 channel In an existing Carrier environment, PSTN traffic through a Class 5 switch Three methods T1 PRI T1 GR303 T1 Dedicated
20
21 PSTN Gateway with PRI s Great way to efficiently use trunk capacity Can oversubscribe T1 links with numbers Bundling multiple PRI s gives fault protection Proven technology Provisioning Asterisk can be complex Dynamic nature of PRI s can add undue provisioning strains with data basing functions
22
23 PSTN Gateway with GR303 Newer protocol, also takes advantage of over subscribing T1 with numbers Bundling multiple T1 s gives fault protection GR303 Aggregator can be used in between Class 5 switch and Asterisk GW to enhance static soft switch provisioning
24 VoIP GW Provisioning Model New services require new processes for adds, moves and changes Automation determines level of human and machine involvement with provisioning Adding VoIP services into existing business processes can be complex Setup I/O database picks Correlating VoIP extensions to telephone # s Configuring features per customer Creating dial plan routing
25 VoIP GW Provisioning Model (cont) Using static VoIP connection channels to PSTN trunks provides minimal impact to daily provisioning tasks When VoIP channels are bonded to Class 5 switch channels, existing provisioning model can facilitate customer turn-up up The VoIP platform can be pre-provisioned provisioned and associated statically to existing databases
26 VoIP GW Provisioning Model (cont) Conceptually, a static soft switch platform can be achieved throughout all VoIP functions and applications CPE can be pre-assigned to customer aggregation channels Customer Aggregation channels can be pre-routed routed to PSTN GW channels Pre-assigned Voice Mail accounts can be associated with each customer channel
27 Application Servers Asterisk is modular software like Linux Asterisk can be setup as monolithic application server for specific task This allows server farms to grow as needed prescribed by the customer or function load per application PSTN GW server can handle 96 channels where the same server may handle 500 Voice Mail channels
28 Customer Aggregation Servers Setup aggregation servers for protocol type, SIP, IAX, MGCP, H323 Firewall each server according to protocol Statically map each CPE channel to a Customer aggregation channel on server Segment Customer IVR or ACD applications on the same server Aggregate multiple Customer groups with different dial plans and call handling routines
29 Dial Plan Servers Provides the core switching unit for the VoIP platform handling all call routing Uses IAX protocol between PSTN GW, Customer Aggregation and Application servers Drops out of call path once transfer is made Dual server, hardware and software fault redundant, self diagnosis for fail over and load sharing Can reload dial plan without restarting service
30
31 Operations and Maintenance 2 maybe 3 VoIP specialist as server quantity increases Create dial plan consistent with best practices of provisioning within host Telco Keep track of software updates that add features or fix bugs Track server usage and resource loading Plan for increasing capacity when needed
32 Software and Hardware Improvements Future plans, look for increase port density for interface cards, i.e. DS3 Look for added protocols, i.e. SS7 Look for VoIP Hardware Vendors adding IAX protocol Look for integration with presence systems, i.e. Jabber, AIM, MSN
33 Conclusion State of technology has never been or will ever be static Don t rely on equipment Vendors to add new products to your service portfolio Telco s pronounce themselves as technology companies, but sell services built by Vendors Embrace the ability to build on a stable and robust VoIP platform, Asterisk
34 Conclusion (cont) Asterisk is a project that delivers into the hands of the novice and experienced alike, a platform for developing telecom applications that bridge the gap between the existing TDM technology and future VoIP technology. Asterisk is well suited for small scale customers and if deployed properly, robust enough to deliver VoIP services to a very, very large customer base.
Leveraging Asterisk to Deliver Large Scale VoIP Services within a Carrier Environment
JR Richardson Engineering for the Masses JR.Richardson@cox.com Leveraging Asterisk to Deliver Large Scale VoIP Services within a Carrier Environment Preface As the voice technology landscape pushes VoIP
More informationAsterisk: A Non-Technical Overview
Asterisk: A Non-Technical Overview Nasser K. Manesh nasser@millenigence.com Millenigence, Inc. 5000 Birch St., Suite 8100 Newport Beach, CA 92660 June 2004, Revised December 2004 Executive Summary Asterisk
More informationGateways and Their Roles
Gateways and Their Roles Understanding Gateways This topic describes the role of voice gateways and their application when connecting VoIP to traditional PSTN and telephony equipment. Analog vs. Digital
More informationehealth 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 informationInternet Telephony Terminology
Internet Telephony Terminology Understanding the business phone system world can be a daunting task to a lay person who just wants a system that serves his or her business needs. The purpose of this paper
More informationWhite Paper. Open Source Telephony: The Evolving Role of Hardware as a Key Enabler of Open Source Telephony in the Business Market.
White Paper Open Source Telephony: The Evolving Role of Hardware as a Key Enabler of Open Source Telephony in the Business Market Produced For: Produced By: July 2006 The Evolving Role of Hardware as a
More informationNEWT Managed PBX A Secure VoIP Architecture Providing Carrier Grade Service
NEWT Managed PBX A Secure VoIP Architecture Providing Carrier Grade Service This document describes the benefits of the NEWT Digital PBX solution with respect to features, hardware partners, architecture,
More informationCisco CME Features and Functionality
Cisco CME Features and Functionality Supported Protocols and Integration Options This topic describes the supported protocols and integration options of Cisco CME. Supported Protocols and Integration FAX
More informationCVOICE 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 informationAn XOP Networks White Paper
Mission Critical Audio Conferencing with VoIP An White Paper i What s Inside EXECUTIVE SUMMARY...2 ISSUES WITH LEGACY MISSION CRITICAL CONFERENCE SYSTEM...3 MONOLITHIC ARCHITECTURE - EXPENSIVE APPROACH
More informationOAISYS and ShoreTel: Call Recording Solution Configuration. An OAISYS White Paper
OAISYS and ShoreTel: Call Recording Solution Configuration An OAISYS White Paper Table of Contents Introduction... 3 ShoreTel Networks... 4 Trunk Side Recording... 4 Station Side Recording... 6 Active
More informationContents. Specialty Answering Service. All rights reserved.
Contents 1 Introduction... 2 2 PBX... 3 3 IP PBX... 4 3.1 How It Works... 4 3.2 Functions of IP PBX... 5 3.3 Benefits of IP PBX... 5 4 Evolution of IP PBX... 6 4.1 Fuelling Factors... 6 4.1.1 Demands from
More informationVoIP-PSTN Interoperability by Asterisk and SS7 Signalling
VoIP-PSTN Interoperability by Asterisk and SS7 Signalling Jan Rudinsky CESNET, z. s. p. o. Zikova 4, 160 00 Praha 6, Czech Republic rudinsky@cesnet.cz Abstract. PSTN, the world's circuit-switched network,
More informationIP- PBX. Functionality Options
IP- PBX Functionality Options With the powerful features integrated in the AtomOS system from AtomAmpd, installing & configuring a cost- effective and extensible VoIP solution is easily possible. 4/26/10
More informationVoice over IP Basics for IT Technicians
Voice over IP Basics for IT Technicians White Paper Executive summary The IP phone is coming or has arrived on desk near you. The IP phone is not a PC, but does have a number of hardware and software elements
More informationCrash Course in Asterisk
Crash Course in Asterisk Despite its name, Asterisk is no mere footnote to the IP-PBX market. The open source product is one of the most disruptive technologies in the industry. Here s what you need to
More informationVoIP from A to Z. NAEO 2009 Conference Cancun, Mexico
VoIP from A to Z NAEO 2009 Conference Cancun, Mexico VoIP glossary What is VoIP? Bandwidth Signaling Codecs Quality of Service (QoS) What is VoIP? Voice over Internet Protocol (VoIP) is the method of transmitting
More informationNetwork Overview. Background Traditional PSTN Equipment CHAPTER
CHAPTER 1 Background Traditional PSTN Equipment Traditional telephone services are engineered and offered over the public switched telephone network (PSTN) via plain old telephone service (POTS) equipment
More informationCVOICE - 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 informationSoftswitch & Asterisk Billing System
Softswitch & Asterisk Billing System IP Telephony Process and architecture is known as Softswitch. Softswitch is used to bridge traditional PSTN and VoIP by linking PSTN to IP networks and managing traffic
More informationOpen source VoIP Networks
Open source VoIP Networks Standard PC hardware inexpensive add-in vs. embedded designs Ing. Bruno Impens Overview History Comparison PC - Embedded More on VoIP VoIP Hardware VoIP more than talk More...
More informationPacketized Telephony Networks
Packetized Telephony Networks Benefits of Packet Telephony Networks Traditionally, the potential savings on long-distance costs was the driving force behind the migration to converged voice and data networks.
More informationAsterisk & ENUM. Extending the Open Source PBX. Michael Haberler, IPA Otmar Lendl, nic.at
Asterisk & ENUM Extending the Open Source PBX Michael Haberler, IPA Otmar Lendl, nic.at Why a ENUM-enable a PBX? your PBX doubles as an IP/PSTN gateway for your existing numbers becomes a dual contact
More informationAT&T SIP Trunk Compatibility Testing for Asterisk
AT&T SIP Trunk Compatibility Testing for Asterisk Mark A. Vince, P.E., AT&T Astricon 2008 September 25, 2008 Phoenix, AZ Agenda Why we tested What we tested Test configuration Asterisk Business Edition
More informationVoice over IP (VoIP) Basics for IT Technicians
Voice over IP (VoIP) Basics for IT Technicians VoIP brings a new environment to the network technician that requires expanded knowledge and tools to deploy and troubleshoot IP phones. This paper provides
More information; Channels 1-8 are incoming voice. ; Channels 13-24 are for data.
Successful businesses usually have the same goal, minimize costs to maximize profits. Today with the plethora of open source solutions, a small business can present a high tech image and still keep a lid
More informationVoIP in the Enterprise
VoIP in the Enterprise Date: March. 2005 Author: Sonia Hanson Version: 1.1 1 1 Background Voice over IP In the late 1990s Voice over IP (VoIP) was seen as a disruptive new technology that had the potential
More informationVoIP Survivor s s Guide
VoIP Survivor s s Guide Can you really save $, improve operations, AND achieve greater security and availability? Presented by Peggy Gritt, Founder and CEO of the VoIP A non-biased organization for the
More informationIP Implementation in Private Branch Exchanges From 9:30 a.m until 4:30 p.m (7 hrs./day) 5 days / week
Course Title: No. of Hours: IP Implementation in Private Branch Exchanges From 9:30 a.m until 4:30 p.m (7 hrs./day) 5 days / week 1 Course Duration: 3 Months (12weeks) No. Of Hours: 7 Hrs./Day- 5 days/week.
More informationImplementing 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 information2- Technical Training (9 weeks) 3- Applied Project (3 weeks) 4- On Job Training (OJT) (4 weeks)
Course Title: Prerequisites: Training Program (5 months) IP Implementation in Private Branch Exchanges Must fresh graduates Communication/Electronics Engineers" 1- Soft Skills Training (4 weeks) 1. Communication
More informationCurso 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 informationConnecting Your Enterprise With Asterisk: IAX to Carriers. Dayton Turner Voxter Communications
Connecting Your Enterprise With Asterisk: IAX to Carriers Dayton Turner Voxter Communications What is IAX? Inter Asterisk exchange Developed by Digium and the Open Source Community Alternative to SIP,
More informationOpen Source Telephony Projects as an Application Development Platform. Frederic Dickey (fdickey@sangoma.com) Director Product Management
Open Source Telephony Projects as an Application Development Platform Frederic Dickey (fdickey@sangoma.com) Director Product Management About this presentation For newcomers to Asterisk For long time CTI
More informationTroubleshooting Voice Over IP with WireShark
Hands-On Course Description Voice over IP is being widely implemented both within companies and across the Internet. The key problems with IP voice services are maintaining the quality of the voice service
More informationOverview of Asterisk (*) Jeff Gunther
Overview of Asterisk (*) Jeff Gunther Agenda Background Introduction to Asterisk and review the core components of it s architecture. Exploration of Asterisk s telephony and call features. Review some
More informationSIP Trunking with Microsoft Office Communication Server 2007 R2
SIP Trunking with Microsoft Office Communication Server 2007 R2 A Dell Technical White Paper By Farrukh Noman Dell Product Group - Enterprise THIS WHITE PAPER IS FOR INFORMATIONAL PURPOSES ONLY, AND MAY
More informationImplementing 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 informationIntegrate VoIP with your existing network
Integrate VoIP with your existing network As organisations increasingly recognise and require the benefits voice over Internet Protocol (VoIP) offers, they stop asking "Why?" and start asking "How?". A
More informationSIP Trunking. Cisco Press. Christina Hattingh Darryl Sladden ATM Zakaria Swapan. 800 East 96th Street Indianapolis, IN 46240
SIP Trunking Christina Hattingh Darryl Sladden ATM Zakaria Swapan Cisco Press 800 East 96th Street Indianapolis, IN 46240 SIP Trunking Contents Introduction xix Part I: From TDM Trunking to SIP Trunking
More informationPETER CUTLER SCOTT PAGE. November 15, 2011
Future of Fax: SIP Trunking PETER CUTLER SCOTT PAGE November 15, 2011 QUESTIONS AND ANSWERS TODAY S SPEAKERS Peter Cutler Vice President of Sales Instant InfoSystems Scott Page Subject Matter Expert Dialogic
More informationPresented by: John Downing, B.Eng, MBA, P.Eng
Presented by: John Downing, B.Eng, MBA, P.Eng John Downing co-founder of TrainingCity. VoIP Training Development Lead. VoIP & SIP Consultant to Telecom & Enterprise Clients. John@TrainingCity.com 613-435-1170
More informationGARTNER REPORT: SIP TRUNKING
GARTNER REPORT: SIP TRUNKING SIP Trunking Slashes U.S. Telecom Expenses by Up to 50%. SUMMARY Network architects and procurement managers can leverage SIP trunking services to slash enterprise telecom
More informationCommonly Supported Fax/Modem Call Flow Configuration Examples
Commonly Supported Fax/Modem Call Flow Configuration Examples Document ID: 115742 Contributed by Hussain Ali, Cisco TAC Engineer. Mar 31, 2014 Contents Introduction Configure Configurations Telco PRI GW
More informationSoftware-Powered VoIP
Software-Powered VoIP Ali Rohani Anthony Murphy Scott Stubberfield Unified Communications Architecture Core Scenarios UC endpoints QOE Monitoring Archiving CDR AOL Public IM Clouds Yahoo Remote Users MSN
More informationAlcatel-Lucent OXO Configuration Guide. For Use with AT&T s IP Flexible Reach Service. Version 1 / Issue 1 Date July 28, 2009
Alcatel-Lucent OmniPCX Office R7.1 Configuration Guide For Use with AT&T s IP Flexible Reach Service Version 1 / Issue 1 Date July 28, 2009 www.alcatel-lucent.com AT&T IP Flexible Reach TABLE OF CONTENTS
More informationThree Network Technologies
Three Network Technologies Network The largest worldwide computer network, specialized for voice ing technique: Circuit-switching Internet The global public information infrastructure for data ing technique:
More informationFig. Setting up of a VoIP call. Fig. Experimental setup
Volume 5, Issue 6, June 2015 ISSN: 2277 128X International Journal of Advanced Research in Computer Science and Software Engineering Research Paper Available online at: www.ijarcsse.com Asterisk VoIP Private
More informationIMPLEMENTING 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 information640-460 - 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 informationBUILDING LARGE CAMPUS ASTERISK-BASED PABX SYSTEMS
BUILDING LARGE CAMPUS ASTERISK-BASED PABX SYSTEMS.0 Background The abundance of Local Area Networks such as those found on most African Universities, and recent advances in Computer and Telephony integration
More informationIntegrating VoIP Phones and IP PBX s with VidyoGateway
Integrating VoIP Phones and IP PBX s with VidyoGateway Updated February 2011 INDEX: I. ABSTRACT.1 II. III. IV. VIDYOGATEWAY OVERVIEW.. 1 NETWORK TOPOLOGIES AND DEFINITIONS...2 CONNECTING TO VIDYOCONFERENCES
More informationWhite Paper: Performance of Host-based Media Processing
White Paper: Performance of Host-based Media Processing The viability of host-based software in providing media processing capabilities continues to expand the possibilities for voice application development.
More informationSIP-Based Solutions in the Contact Center: Using Dialogic Media Gateways with the Genesys Voice Platform
-Based Solutions in the Contact Center: To stay competitive and keep their customers happy and loyal, companies are working hard to enhance customer service as costeffectively as possible. Contact centers
More informationAT&T IP Flexible Reach Service
I. Service Overview II. Service Components, standard and options I. Service Overview AT&T s Business Voice over IP ( AT&T BVoIP ) portfolio of services enable the transmission of voice telephone calls
More informationVitalPBX. Hosted Voice That Works. For You
VitalPBX Hosted Voice That Works For You Vital Voice & Data s VitalPBX VVD Hosted PBX solutions provide you with the stability of a traditional telephone PBX system and the flexibility that only a next
More informationAllstream Converged IP Telephony
Allstream Converged IP Telephony SIP Trunking Solution An Allstream White Paper 1 Table of contents Introduction 1 Traditional trunking: a quick overview 1 SIP trunking: a quick overview 1 Why SIP trunking?
More informationIP PBX using SIP. Voice over Internet Protocol
IP PBX using SIP Voice over Internet Protocol Key Components for an IP PBX setup Wireless/Fiber IP Networks (Point to point/multi point, LAN/WAN/Internet) Central or Multicast SIP Proxy/Server based Virtual
More informationCopyright and Trademark Statement
Contents VoIP Starts with SmartNode...3 Why SmartNode?...3 SmartNode Product Comparison...5 VoIP Appliance with Embedded Windows...7 Carrier-Grade TDM + VoIP SmartMedia Gateways...8 Enterprise Solutions...9
More informationB rismark. Open Source IP PBX The Future of Telephony. T: +92.21.111199299 W: www.birsmark.com
Open Source IP PBX The Future of Telephony What is Asterisk Asterisk is the world s leading open source telephony engine and tool kit. Offering flexibility unheard of in the world of proprietary communications,
More informationAvaya Call Recording Solution Configuration
Avaya Call Recording Solution Configuration Avaya IP Office Americas Headquarters OAISYS 7965 South Priest Drive, Suite 105 Tempe, AZ 85284 USA www.oaisys.com (480) 496-9040 CONTENTS 1 Introduction 2 Overview
More informationVoIP and IP Telephony @ IT Tralee
VoIP and IP Telephony @ IT Tralee chris.bradshaw@staff.ittralee.ie Presentation outline: Basic overview of IP telephony and technology Detailed overview of VoIP @ IT Tralee deployment How IPT has benefited
More informationSelecting the Right SIP Phone for Your IP PBX By Gary Audin May 5, 2014
Selecting the Right SIP Phone for Your IP PBX By Gary Audin May 5, 2014 There are many Session Initiation Protocol (SIP) phones on the market manufactured by IP PBX vendors and third parties. Selecting
More informationConverged Telephony Solution. Technical White Paper
CTS White Paper Page 1 of 11 Converged Telephony Solution Technical White Paper ٠ May 2004 CTS White Paper Page 2 of 11 Converged Telephony Solution White Paper The focus of this white paper is to explain
More information640-460. 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 informationAdvanced LCR (Least Cost Router) With SIP Proxy Server
With SIP Proxy Server It s all about Reducing Cost!!! WHY ADVANCED LCR (Least Cost Routing) Advanced LCR is a product by Advanced Communications; the same parent company of AdvancedVoIP.com, a widely used
More informationUsing Asterisk with Odin s OTX Boards
Using Asterisk with Odin s OTX Boards Table of Contents: Abstract...1 Overview...1 Features...2 Conclusion...5 About Odin TeleSystems Inc...5 HeadQuarters:...6 Abstract Odin TeleSystems supports corporate
More informationMediatrix 3000 with Asterisk June 22, 2011
Mediatrix 3000 with Asterisk June 22, 2011 Proprietary 2011 Media5 Corporation Table of Contents Introduction... 3 Network Topology... 3 Equipment Detail... 3 Configuration of the Fax Extension... 4 Configuration
More informationA Cable Telephony Case Study: HOT Telecom
A Cable Telephony Case Study: HOT Telecom Powered by AudioCodes Mediant 2000 and Mediant 8000 Media Gateways HOT Telecom, the Israeli cable operator has deployed AudioCodes Media Gateways into their VoIP
More informationSpecialty Answering Service. All rights reserved.
0 Contents 1 Introduction... 3 2 Features... 4 2.1 Hardware Requirement... 4 2.2 Protocol Support... 4 2.3 Configuration... 4 2.4 Applications... 5 2.5 Graphical User Interfaces... 5 3 History and Evolution
More informationAsterisk Calling Card & Billing System
Asterisk Calling Card & Billing System Asterisk based Calling Card Billing (A2Billing), PC to Phone Billing, IPPhone to Phone Billing with Admin Module, Reseller Module, Customer Module & Account (PIN)
More informationEnterprise open source VoIP with Asterisk
1 of 7 2/25/2008 9:32 AM On CNET: 6 Worst MP3 players of 2007 BNET Business Network: BNET TechRepublic ZDNet Enterprise open source VoIP with Asterisk by George Ou Jun 29, 2006 7:00:00 AM Takeaway: A lot
More informationIP Telephony with Asterisk. Sunday A. Folayan
IP Telephony with Asterisk Sunday A. Folayan There lived the PSTN. A few years ago, everyone struggled to convert data (IP) into sound, and move it over the Public Switched Telephone Network (PSTN) infrastructure
More informationK-Net VoIP telephone system May 31, 2005
K-Net VoIP telephone system May 31, 2005 Description. The K-Net VoIP telephone system is an internal telephone system that uses the existing K-Net data network. Priority is given to the VoIP traffic to
More informationTelco Carrier xsps Solutions. www.first.gr
Telco Carrier xsps Solutions www.first.gr TELCO CORE INFRUSTRUCTURE There are some necessities in the Telco world...to own the best available infrastructure at the most affordable cost. First Telecom s
More informationEnabling Innovation - Unleashing Unified Communications: Best Practices and Case Studies. October 18-19, 2011
Enabling Innovation - Unleashing Unified Communications: Best Practices and Case Studies Grant Bykowy, Director Voice & IP Communications Product Management & Marketing October 18-19, 2011 How can the
More informationChoosing a Dialogic Product Option for Creating a PSTN-HMP Interface
Whitepaper PSTN-HMP Interface Options Choosing a Dialogic Product Option for Creating a PSTN-HMP Interface Environment Helps Determine Product Choice for TDM-IP Hybrid Media Server System with Dialogic
More informationConnect your Control Desk to the SIP world
Connect your Control Desk to the SIP world Systems in
More information"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 informationImplementing 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 informationCisco AS5400 Series Universal Gateways How to Order a Cisco AS5400XM Universal Gateway
Ordering Guide Cisco AS5400 Series Universal Gateways How to Order a Cisco AS5400XM Universal Gateway PRODUCT OVERVIEW Cisco AS5400XM Universal Gateways offer unparalleled capacity and the flexibility
More information1. Mobile VoIP solutions and Services:
About Us We at Mir Technologies do 3 things. 1. Mobile VoIP solutions and Services: mtel is an International Brand. mtel created partnerships with esteemed global VoIP operators through a focused and stable
More informationAudioCodes Gateway in the Lync Environment
AudioCodes Gateway in the Lync Environment Course: Audience: Prerequisites: Products: Four days hands-on, technical instruction covering installation, configuration, maintenance, troubleshooting and administration
More informationBuilding Voice VPN with Simton IPX
Building Voice VPN with Simton IPX (Simton Technologies, Inc.) Version 6 With Simton IPX, the small and medium businesses can easily consolidate data and voice network together to increase productivity,
More informationBasics of VoIP Termination
Basics of VoIP Termination Version 1.1 July 26, 2006 AdvancedVoIP.com sales@advancedvoip.com support@advancedvoip.com Phone: +1 213 341 1431 Copyright AdvancedVoIP.com, 1999-2006. All Rights Reserved.
More informationIntegrated Communications Platform
Integrated Communications Platform Voice services Data networking Applications Remote administration and multisite management The InstantOffice system is a robust, modular Integrated Communications Platform
More informationEnterprise 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 informationSIP Trunking DEEP DIVE: The Service Provider
SIP Trunking DEEP DIVE: The Service Provider Larry Keefer, AT&T Consulting UC Practice Director August 12, 2014 2014 AT&T Intellectual Property. All rights reserved. AT&T, the AT&T logo and all other AT&T
More informationIntroducing 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 informationVoice over IP Technologies
Voice over IP Technologies Voice Over IP Overview VoIP is an emerging technology that allows voice calls to be made over an IP network. Vendors have been pushing VoIP for a few years, but many potential
More informationSIP Trunking: The New Normal in the Cloud Era
SIP Trunking: The New Normal in the Cloud Era Executive Summary As IP and VoIP technologies continue to mature and the thirst for cloud services escalates, the SIP Trunking market is growing rapidly, driven
More informationWhich VoIP Architecture Makes Sense For Your Contact Center?
a White Paper from Vanguard Communications Which VoIP Architecture Makes Sense For Your Contact Center? by Areg Gharakhanian August 2002 Vanguard Communications Corporation 100 American Road Morris Plains,
More informationTotal Recall Max SIP VoIP Call Recording Server
Total Recall Max SIP VoIP Call Recording Server Introduction In an increasingly security conscious, results driven and litigious world, communications recording is vital to meeting your duty of care, management
More informationApplication Note. Using a Dialogic Media Gateway Series as a PSTN Gateway with an Asterisk IP-PBX Server
Using a Dialogic Media Gateway Series as a PSTN Gateway with an Asterisk IP-PBX Server Using a Dialogic Media Gateway Series as a PSTN Gateway with an Asterisk IP-PBX Server Executive Summary This application
More informationand Voice Applications Eyal Wirsansky, Verso Technologies JaxJUG
Voice Over IP, and Voice Applications Eyal Wirsansky, Verso Technologies JaxJUG Analog Telephony Mr. W AG Bell X What the *!@# is aa Switch?? Moving to Digital Voice (TDM) Separation of Voice and Signaling
More informationADVOSS SIP APPLICATION SERVERS
ADVOSS SIP APPLICATION SERVERS PRODUCT DATA SHEET COPYRIGHT ADVOSS.COM, 2007 2011 ALL RIGHTS RESERVED This document is property of AdvOSS Page 1 TABLE OF CONTENTS 1 AdvOSS SIP Application Servers... 3
More informationSTUDY PAPER on IP PABX. (IP based PRIVATE AUTOMATIC BRANCH EXCHANGE) Table of Contents. 1. Introduction... 2. 2. History & Evolution of PABX...
STUDY PAPER on IP PABX (IP based PRIVATE AUTOMATIC BRANCH EXCHANGE) Table of Contents 1. Introduction... 2 2. History & Evolution of PABX... 2 3. The Basic Architecture, its Functions and Interfaces...
More informationReplixFax Fax over IP (FoIP) Technical Overview and Benefits
ReplixFax Fax over IP (FoIP) Technical Overview and Benefits www.softlinx.com Table of Contents INTRODUCTION... 3 HOW FAX OVER IP (FOIP) WORKS... 3 FAX PROTOCOLS FOR FOIP... 3 TRADITIONAL FAX SERVER VS.
More informationImplementing Cisco IP Telephony & Video, Part 1 CIPTV1 v1.0; 5 Days; Instructor-led
Implementing Cisco IP Telephony & Video, Part 1 CIPTV1 v1.0; 5 Days; Instructor-led Course Description Implementing Cisco IP Telephony & Video, Part 1 (CIPTV1) v1.0 is a five-day course that prepares the
More informationIP 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