Frequently Asked Questions about Integrated Access

Size: px
Start display at page:

Download "Frequently Asked Questions about Integrated Access"

Transcription

1 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 federal government. These LATAs determine local calling areas. A call is considered local if it is an intralata call (within your local access transport area) and within your state. Calls outside your LATA (interlata calls) or outside your state are considered long distance. U.S. long distance calling includes all 50 states. Calls outside of the U.S. are billed using our international calling rates. Calls to Canada and Puerto Rico are considered international. What are the rates for local, long distance, and international calls? Local calls There is no charge for local calls. Long distance U.S. long distance calls are currently charged at 4 cents per minute or select a calling plan with 2,000, 5,000, 10,000, or 18,000 monthly long distance minutes (rates as low as 2.2 cents per minute). Excess minutes are charged at 4 cents per minute. Unused minutes from one month do not "roll over" to the next month. International calls The international rate sheet is available at Toll-free calls Inbound toll-free calls from the U.S. and Canada are currently charged at 4 cents per minute. Covad toll-free numbers do not accept calls from outside the U.S. and Canada. Is faxing supported? Covad Integrated Access can support fax transmissions of up to three 20-page documents per hour. Companies with heavy fax requirements should maintain a separate stand-alone analog phone line for faxing. Fax machines should be set at 9600 BAUD for optimal results. Customers should refer to their fax machine manual for instructions on changing the BAUD rate. Covad Integrated Access service uses the G.711 codec and does not support the T.38 protocol. What if I exceed the recommended faxing limits? Covad has successfully tested fax transmissions of 20 pages at a frequency of three times an hour. It is possible that higher usage may work, but it has not been tested. If you exceed the recommended faxing limit, it is possible that a fax may fail to complete properly or extra image lines may be added to the transmission. The more pages in a faxed document, and the more frequently documents are faxed, the greater the likelihood that this will occur. Why are there issues with faxing over IP networks? Fax transmissions over IP networks are not very tolerant of latency or congestion. Fax packets over an IP network use UDP and are not automatically re-transmitted if there are errors. Missing packets can sometimes cause the fax to fail completely. If this happens, the customer can simply re-fax it. Covad is a MegaPath brand. For Customer Support, call MEGA 1

2 Are alarm systems or point-of-sale (POS) systems supported? No. Covad strongly recommends that customers maintain a separate analog phone line for point-of-sale systems and alarm systems. Can our company use an auto dialer or predictive dialer with the service? No. The use of auto dialers or predictive dialers is prohibited with Covad Integrated Access service. Does the Covad Integrated Access solution have the ability to block or restrict incoming or outgoing calls? No. Integrated Access service does not block incoming or outgoing calls on specific phone lines or phone numbers, but this feature may be available through your PBX system. Please note that Integrated Access automatically blocks outgoing calls from your toll-free phone numbers only calls to 911 are allowed from a toll-free number. Equipment What PBX systems do you support? The Covad Integrated Access service supports a variety of analog, digital, and IP-PBXs. The service will work with most analog and digital PBX systems, and a fast-growing list of IP-PBX systems. What is the supported codec for the Covad Integrated Access service? Covad uses the G.711 codec to help ensure optimal voice quality. How much bandwidth does the G.711 codec use? The G.711 uses pulse code modulation (PCM) of voice frequencies on a 64 Kbps channel and consumes approximately 106 Kbps per voice call. What type of equipment is provided with the Covad Integrated Access service? Covad provides the Cisco 2431 and Samsung IBG 1003 Integrated Access Devices (IADs) if you have an analog system or a digital PBX. We supply a Cisco 1841 or a Samsung IBG 1000 T1 router if you have an IP-PBX. IP-PBX Equipment Do I have to use the Network Time Protocol (NTP) server in the IP-PBX configuration? Yes. The NTP server must be configured in the IP-PBX to keep the date/time in sync between the Covad voice network and the IP-PBX so that phone usage is accurately reflected. Covad is a MegaPath brand. For Customer Support, call MEGA 2

3 Support Who do I contact when I have questions about my Integrated Access service once it is installed? Call MegaPath Customer Support at MEGA. We can help you make changes to your service, handle billing issues, answer technical questions, and more. What do I do if I need to move my office location, add phone lines, or make other changes to my service after it's installed? If you need to make changes or additions to your service after it is operational, contact MegaPath at , option 1. Who do I contact if I have a billing issue? Call MegaPath at , option 5, for assistance. Can I pay my monthly bill by credit card? You can change your billing option from invoice to credit card or electronic funds transfer by logging into your MegaPath Customer Portal. Can our company change the phone number listed on our Caller ID? Yes. Contact MegaPath at , option 1. How can I add another business location to my existing Integrated Access service? Contact MegaPath at , option 1. What if I'm moving my office location? What happens to my Integrated Access service? Call MegaPath at , option 1, to verify that Integrated Access service is available at your new location. The customer support representative can handle the details of moving your Integrated Access service. Technical Details How are phone numbers treated with an analog PBX or key telephone system? Covad Integrated Access supports a standard dial-peer plan. All numbers are shared. Incoming calls on any company phone number will ring on the first available phone. Each user must have an extension programmed by the key system or PBX. What is my network layout with Integrated Access added to it? Covad provides equipment (either an IAD or a router, depending on your PBX equipment) to connect your network to the Covad network. The IAD or router is configured to support two separate subnets in your environment one for voice and one for data. Covad is a MegaPath brand. For Customer Support, call MEGA 3

4 Is there any inter-routing between the voice and data subnets? No. There is no inter-routing between the two subnets, because this would allow data transfers to potentially affect the quality of the voice service. Can I request that Covad set up the router so that all traffic goes over a single interface (commonly referred to as 802.1Q trunking)? No. This is not a supported configuration. What VoIP protocols does Covad use? The Covad Integrated Access service supports only session initiation protocol (SIP) as the signaling protocol to access and utilize the trunks for termination to and from the public switched telephone network (PSTN). How many public IP addresses are provided? You can choose the number of IP addresses you need for your situation. For example, if you request 8 IPs, Covad will provide you with 5 usable IP addresses for your data subnet. If you have an analog or digital PBX system, Covad will assign an additional public IP address to your integrated access device (IAD). If you have an IP-PBX, Covad provides 13 usable IP addresses for your voice subnet. If your IP-PBX does not function as an application layer gateway (ALG), you will need to assign one IP address for each phone on your voice subnet. Check with your PBX vendor to determine if your IP-PBX utilizes ALG. If you need additional IP addresses, you can request them when you place your order. An IP address justification form will be required. Are the IP settings on the data and voice network interfaces static or DHCP? The IP addresses can be acquired via DHCP or can be specifically assigned from the range of public IP addresses that Covad allotted to you. If you have an IP-PBX, Covad recommends that you specifically assign one of your IP addresses to it. Addresses for the phones can either be acquired through DHCP or specifically designated, whichever you prefer. Please note that if you choose to assign static IP addresses to the phones, you will also have to assign static IP addresses to the DNS server and NTP server. How much bandwidth is available for phone calls at any given time? Integrated Access phone service capacity is defined by the number of phone lines you purchase as part of your Integrated Access service. If someone calls your business when all your voice bandwidth is in use, the caller will hear a busy signal. Callers who dial out when you are using all your voice bandwidth will hear a tone indicating they should try their call again later. When you are using less bandwidth for phone calls, Integrated Access dynamically allocates the extra bandwidth to data traffic until it is needed again for phone calls. How much bandwidth does an internal call take up on the T1 circuit? If the internal call is within the same building (i.e., phones behind one router), the phone conversation doesn't use any circuit bandwidth. The initial signal setup, as the call is dialed, uses a minimal amount of bandwidth as the voice switch in our core network prepares for action. But after the conversation starts, the packets stay within the building. So, if two cube Covad is a MegaPath brand. For Customer Support, call MEGA 4

5 neighbors are talking to one another and the circuit goes down, their conversation will remain in progress. How does Covad handle security on voice calls? Security concerns can be divided into three areas: 1. Security of the phone conversation 2. Security of the end point devices (from hacker attacks) 3. Security of the data network (so voice is not used as a back door to get into the data network) Covad voice traffic is always carried over our private network, not the public Internet. By using this approach, we provide a secure conversation because we restrict access to our network at all times. Local, long distance, and international voice packets generated from a phone at any of our sites traverse the point-to-point T1 line directly to the Covad voice service office (VSO) and are handed off to a carrier who in turn hands the voice traffic to the public switched telephone network (PSTN). These packets never touch the Internet. This is not always the case with other voice over IP (VoIP) providers. By keeping voice traffic on our private network, we also make it harder for anyone to reach end point devices such as IP phones, routers, or switches. We manage the traffic that goes to the end points and ensure that only voice traffic that we can recognize gets to those end points. This helps fend off end point attacks. At the Covad VSO, unrequested voice traffic is blocked at the gateway. Because we allow only legitimate voice traffic to reach the end points, it makes it very hard to use the end point as a back door to the data network. In addition, Covad configures two separate subnets at your location one for voice and one for data. This helps ensure that issues can be isolated if a problem is encountered. Terminology What is voice-optimized access (VOA)? Covad uses the term voice-optimized access (VOA) for the technology we use to prioritize voice traffic over data traffic on the Covad network. This helps ensure superb voice quality all the way from your phone to the public switched telephone network (PSTN). What is dynamic bandwidth allocation? Dynamic bandwidth allocation provides automated bandwidth management so that the traffic on a single broadband link is allocated to voice or data as needed. Because most users do not transmit high volumes of data at all times and because voice traffic is more sensitive to delays than data traffic, dynamic bandwidth allocation allows inbound and outbound phone calls to take priority over data traffic, assuring consistently high voice quality and providing you more value from a single T1 connection. What is a dial peer? A dial peer is another name for a call end point or destination, usually a phone. An end point can be any device that can originate or receive a call. Covad is a MegaPath brand. For Customer Support, call MEGA 5

6 What is a codec? A codec (compressor/decompressor) is a program or piece of equipment that can compress signals, such as voice and video, into a digital data stream for transmission. Codecs are used because a compressed file takes up less storage space and can be transferred across a network more quickly and smoothly. What is SIP? SIP stands for Session Initiation Protocol. It is a signaling protocol used for establishing sessions in an IP network. SIP creates, modifies, and terminates "sessions" with one or more participants. A session could be a simple two-way telephone call or a collaborative multimedia conference session. SIP invitations are used to create sessions and carry information that allows participants to agree on a set of compatible media types. SIP makes use of elements called proxy servers to help route requests to users' current locations, authenticate and authorize users for services, implement provider call-routing policies, and provide features to users. SIP also provides a registration function that allows users to upload their current locations to the proxy servers. SIP is the protocol of choice for signaling for VoIP traffic. It establishes and terminates voice "calls" flexibly and efficiently and minimizes hardware requirements and recurring network charges associated with traditional phone service, thus reducing costs while maintaining features and quality. What is SIP trunking? A SIP trunk is a logical connection between an IP-PBX and a service provider's application servers that allows VoIP traffic to be exchanged between the two. When a call is placed from an internal phone to an external number, the PBX sends the necessary information to the SIP trunk provider, who establishes the call to the dialed number and acts as an intermediary for the call. All signaling and voice traffic between the PBX and the provider is exchanged using SIP and RTP protocol packets over the IP network. What is QoS? Quality of Service. It refers to giving some network traffic higher priority. QoS is based on the fact that some traffic is more resilient. Voice traffic is not resilient because it is time sensitive. Ideally, time-sensitive traffic (like phone calls) should be given priority over less sensitive traffic (like ) to ensure the timely delivery and quality of the voice conversation. To implement QoS, you must first classify what high-priority traffic is, then handle that traffic better than the low-priority traffic. There are many companies that cannot provide QoS in their voice network. Covad does, and we do it throughout the entire network, from customer phone to our switch and onto the public switched telephone network (PSTN). Covad is a MegaPath brand. For Customer Support, call MEGA 6

Converged Voice Service Summary

Converged Voice Service Summary SERVICE DELIVERY OVERVIEW Zayo Enterprise Networks (ZEN) offers a managed converged voice and Internet solution for businesses owning and managing a premise-based KTS or PBX system. Voice and Internet

More information

SIP Trunking Quick Reference Document

SIP Trunking Quick Reference Document SIP Trunking Quick Reference Document Publication Information SAMSUNG TELECOMMUNICATIONS AMERICA reserves the right without prior notice to revise information in this publication for any reason. SAMSUNG

More information

ATA: An Analogue Telephone Adapter is used to connect a standard telephone to a high-speed modem to facilitate VoIP and/or calls over the Internet.

ATA: An Analogue Telephone Adapter is used to connect a standard telephone to a high-speed modem to facilitate VoIP and/or calls over the Internet. KEY VOIP TERMS 1 ACD: Automatic Call Distribution is a system used to determine how incoming calls are routed. When the ACD system receives an incoming call it follows user-defined specifications as to

More information

nexvortex SIP Trunking Implementation & Planning Guide V1.5

nexvortex SIP Trunking Implementation & Planning Guide V1.5 nexvortex SIP Trunking Implementation & Planning Guide V1.5 510 S PRING S TREET H ERNDON VA 20170 +1 855.639.8888 Introduction Welcome to nexvortex! This document is intended for nexvortex Customers and

More information

Skype Connect Requirements Guide

Skype Connect Requirements Guide Skype Connect Requirements Guide Version 4.0 Copyright Skype Limited 2011 Thinking about implementing Skype Connect? Read this guide first. Skype Connect provides connectivity between your business and

More information

ACD: Average Call Duration is the average duration of the calls routed bya a VoIP provider. It is a quality parameter given by the VoIP providers.

ACD: Average Call Duration is the average duration of the calls routed bya a VoIP provider. It is a quality parameter given by the VoIP providers. ACD: Average Call Duration is the average duration of the calls routed bya a VoIP provider. It is a quality parameter given by the VoIP providers. API: An application programming interface (API) is a source

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

Integrate VoIP with your existing network

Integrate 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 information

Application Note Configuring the Synapse SB67070 SIP Gateway for Broadvox GO! SIP Trunking

Application Note Configuring the Synapse SB67070 SIP Gateway for Broadvox GO! SIP Trunking Configuring the Synapse SB67070 SIP Gateway for Broadvox GO! SIP Trunking 2012 Advanced American Telephones. All Rights Reserved. AT&T and the AT&T logo are trademarks of AT&T Intellectual Property licensed

More information

Glossary of Telco Terms

Glossary of Telco Terms Glossary of Telco Terms Access Generally refers to the connection between your business and the public phone network, or between your business and another dedicated location. A large portion of your business

More information

SIP Trunking. Service Guide. www.megapath.com. Learn More: Call us at 877.634.2728.

SIP Trunking. Service Guide. www.megapath.com. Learn More: Call us at 877.634.2728. Service Guide Learn More: Call us at 877.634.2728. www.megapath.com What is MegaPath SIP Trunking? SIP Trunking enables your business to reduce costs and simplify IT management by combining voice and Internet

More information

AT&T IP Flex Reach/ IP Toll Free Configuration Guide IC 3.0 with Interaction SIP Proxy

AT&T IP Flex Reach/ IP Toll Free Configuration Guide IC 3.0 with Interaction SIP Proxy INTERACTIVE INTELLIGENCE AT&T IP Flex Reach/ IP Toll Free Configuration Guide IC 3.0 with Interaction SIP Proxy Version 1.7 9/2/2009 TABLE OF CONTENTS 1 AT&T... 5 1.1 Introduction... 5 1.2 Product Descriptions...

More information

Integrated Voice. Service Guide. www.megapath.com. Learn More: Call us at 877.634.2728. Save money and maximize bandwidth efficiency

Integrated Voice. Service Guide. www.megapath.com. Learn More: Call us at 877.634.2728. Save money and maximize bandwidth efficiency Service Guide Save money and maximize bandwidth efficiency Learn More: Call us at 877.634.2728. www.megapath.com Table of Contents Product overview... 3 What is integrated voice?... 3 How it works... 4

More information

MINIMUM NETWORK REQUIREMENTS 1. REQUIREMENTS SUMMARY... 1

MINIMUM NETWORK REQUIREMENTS 1. REQUIREMENTS SUMMARY... 1 Table of Contents 1. REQUIREMENTS SUMMARY... 1 2. REQUIREMENTS DETAIL... 2 2.1 DHCP SERVER... 2 2.2 DNS SERVER... 2 2.3 FIREWALLS... 3 2.4 NETWORK ADDRESS TRANSLATION... 4 2.5 APPLICATION LAYER GATEWAY...

More information

Configuring a Mediatrix 500 / 600 Enterprise SIP Trunk SBC June 28, 2011

Configuring a Mediatrix 500 / 600 Enterprise SIP Trunk SBC June 28, 2011 Configuring a Mediatrix 500 / 600 Enterprise SIP Trunk SBC June 28, 2011 Proprietary 2011 Media5 Corporation Table of Contents Introduction... 3 Solution Overview... 3 Network Topology... 4 Network Configuration...

More information

Configuration Guide for connecting the Eircom Advantage 4800/1500/1200 PBXs to the Eircom SIP Voice platform.

Configuration Guide for connecting the Eircom Advantage 4800/1500/1200 PBXs to the Eircom SIP Voice platform. Configuration Guide for connecting the Eircom Advantage 4800/1500/1200 PBXs to the Eircom SIP Voice platform. 1 Contents Introduction.... 3 Installing the Applications Module... 4 Ordering a Licence for

More information

Voice over IP Basics for IT Technicians

Voice 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 information

EarthLink Business SIP Trunking. NEC SV8300 IP PBX Customer Configuration Guide

EarthLink Business SIP Trunking. NEC SV8300 IP PBX Customer Configuration Guide EarthLink Business SIP Trunking NEC SV8300 IP PBX Customer Configuration Guide Publication History First Release: Version 1.0 May 18, 2012 CHANGE HISTORY Version Date Change Details Changed By 1.0 5/18/2012

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

How to Configure the Toshiba Strata CIX for use with Integra Telecom SIP Solutions

How to Configure the Toshiba Strata CIX for use with Integra Telecom SIP Solutions How to Configure the Toshiba Strata CIX for use with Integra Telecom SIP Solutions Overview: This document provides a reference for configuration of the Toshiba Strata CIX IP PBX to connect to Integra

More information

EarthLink Business SIP Trunking. NEC SV8100 IP PBX Customer Configuration Guide

EarthLink Business SIP Trunking. NEC SV8100 IP PBX Customer Configuration Guide EarthLink Business SIP Trunking NEC SV8100 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

SIP Trunking Application Notes V1.3

SIP Trunking Application Notes V1.3 SIP Trunking Application Notes V1.3 Publication Information SAMSUNG TELECOMMUNICATIONS AMERICA reserves the right without prior notice to revise information in this publication for any reason. SAMSUNG

More information

TELEPHONE MAN OF AMERICA. Earning Your Business Every Step of the Way!

TELEPHONE MAN OF AMERICA. Earning Your Business Every Step of the Way! TELEPHONE MAN OF AMERICA Earning Your Business Every Step of the Way! Specializing in Telecom Equipment of all Brands, Carrier Services, Technician Services, Maintenance Agreements & Purchasing Excess

More information

SIP Trunking and Voice over IP

SIP Trunking and Voice over IP SIP Trunking and Voice over IP Agenda What is SIP Trunking? SIP Signaling How is Voice encoded and transported? What are the Voice over IP Impairments? How is Voice Quality measured? VoIP Technology Confidential

More information

How to Configure the NEC SV8100 for use with Integra Telecom SIP Solutions

How to Configure the NEC SV8100 for use with Integra Telecom SIP Solutions How to Configure the NEC SV8100 for use with Integra Telecom SIP Solutions Overview: This document provides a reference for configuration of the NEC SV8100 IP PBX to connect to Integra Telecom SIP trunks.

More information

Requirements of Voice in an IP Internetwork

Requirements of Voice in an IP Internetwork Requirements of Voice in an IP Internetwork Real-Time Voice in a Best-Effort IP Internetwork This topic lists problems associated with implementation of real-time voice traffic in a best-effort IP internetwork.

More information

VoIP CONFIGURATION GUIDE FOR MULTI-LOCATION NETWORKS

VoIP CONFIGURATION GUIDE FOR MULTI-LOCATION NETWORKS VoIP CONFIGURATION GUIDE FOR MULTI-LOCATION NETWORKS INTRODUCTION About this guide This guide is designed to help you plan and configure a TalkSwitch multi-location network for Voice over IP (VoIP). NOTE:

More information

Application Notes for Configuring Cablevision Optimum Voice SIP Trunking with Avaya IP Office - Issue 1.1

Application Notes for Configuring Cablevision Optimum Voice SIP Trunking with Avaya IP Office - Issue 1.1 Avaya Solution & Interoperability Test Lab Application Notes for Configuring Cablevision Optimum Voice SIP Trunking with Avaya IP Office - Issue 1.1 Abstract These Application Notes describe the procedures

More information

Application Notes for Avaya IP Office 7.0 Integration with Skype Connect R2.0 Issue 1.0

Application Notes for Avaya IP Office 7.0 Integration with Skype Connect R2.0 Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Avaya IP Office 7.0 Integration with Skype Connect R2.0 Issue 1.0 Abstract These Application Notes describe the steps to configure an Avaya

More information

Fonality. Optimum Business Trunking and the Fonality Trixbox Pro IP PBX Standard Edition V4.1.2- p13 Configuration Guide

Fonality. Optimum Business Trunking and the Fonality Trixbox Pro IP PBX Standard Edition V4.1.2- p13 Configuration Guide Fonality Optimum Business Trunking and the Fonality Trixbox Pro IP PBX Standard Edition V4.1.2- p13 Configuration Guide Fonality Table of Contents 1. Overview 2. SIP Trunk Adaptor Set-up Instructions 3.

More information

General Guidelines for SIP Trunking Installations

General Guidelines for SIP Trunking Installations SIP Trunking Installations General Guidelines for SIP Trunking Installations 1) How do I setup my SIP trunk for inbound/outbound calling? We authenticate IP-PBX SIP Trunking traffic by: IP Authentication

More information

SIP Domain/Proxy, Ring Detect Extension or/and Page Audio Extension, (The 8180 needs its own phone extension) Authentication ID, Password,

SIP Domain/Proxy, Ring Detect Extension or/and Page Audio Extension, (The 8180 needs its own phone extension) Authentication ID, Password, The 8180 requires the information below: SIP Domain/Proxy, Ring Detect Extension or/and Page Audio Extension, (The 8180 needs its own phone extension) Authentication ID, Password, Get TalkSwitch SIP Domain/Proxy:

More information

Troubleshooting Voice Over IP with WireShark

Troubleshooting 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 information

Voice over IP (VoIP) Basics for IT Technicians

Voice 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

EarthLink Business SIP Trunking. Switchvox SMB 5.5 & Adtran SIP Proxy Implementation Guide

EarthLink Business SIP Trunking. Switchvox SMB 5.5 & Adtran SIP Proxy Implementation Guide EarthLink Business SIP Trunking Switchvox SMB 5.5 & Adtran SIP Proxy Implementation Guide Publication History First Release: Version 1.0 April 20, 2012 CHANGE HISTORY Version Date Change Details Changed

More information

OfficeMaster Gate (Virtual) Enterprise Session Border Controller for Microsoft Lync Server. Quick Start Guide

OfficeMaster Gate (Virtual) Enterprise Session Border Controller for Microsoft Lync Server. Quick Start Guide OfficeMaster Gate (Virtual) Enterprise Session Border Controller for Microsoft Lync Server Quick Start Guide October 2013 Copyright and Legal Notice. All rights reserved. No part of this document may be

More information

VOIP NETWORK CONFIGURATION GUIDE RELEASE 6.10

VOIP NETWORK CONFIGURATION GUIDE RELEASE 6.10 TALKSWITCH DOCUMENTATION VOIP NETWORK CONFIGURATION GUIDE RELEASE 6.10 CT.TS005.002606 ANSWERS WITH INTELLIGENCE INTRODUCTION About this guide This guide will help you plan and configure a TalkSwitch system

More information

Broadvox SIP Trunking. Frequently Asked Questions (FAQs)

Broadvox SIP Trunking. Frequently Asked Questions (FAQs) Broadvox SIP Trunking Frequently Asked Questions (FAQs) Table of Contents Can a Call Center with an automated dialer use Broadvox services? 3 Can I connect to Broadvox services if I have a dynamic IP address?

More information

Integrated Voice. Service Guide. Save money and maximize bandwidth efficiency Version 201009

Integrated Voice. Service Guide. Save money and maximize bandwidth efficiency Version 201009 Integrated Voice Service Guide Save money and maximize bandwidth efficiency Version 201009 TABLE OF CONTENTS TABLE OF CONTENTS...2 PRODUCT OVERVIEW...3 WHAT IS INTEGRATED VOICE?...3 HOW IT WORKS...4 BASE

More information

General Guidelines for SIP Trunking Installations

General Guidelines for SIP Trunking Installations General Guidelines for SIP Trunking Installations 1) How do I setup my SIP trunk for inbound/outbound calling? We authenticate IP-PBX SIP Trunking traffic by: IP Authentication (IP address) or Digest Authentication

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

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

Internet Telephony Terminology

Internet 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 information

Tech Bulletin 2012-002. IPitomy AccessLine SIP Provider Configuration

Tech Bulletin 2012-002. IPitomy AccessLine SIP Provider Configuration support@ipitomy.com 941.306.2200 Tech Bulletin 2012-002 Description This guide is intended to streamline the installation of AccessLine SIP trunks in the IPitomy IP PBX. In our combined testing we determined

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

How to Configure the Avaya IP Office 6.1 for use with Integra Telecom SIP Solutions

How to Configure the Avaya IP Office 6.1 for use with Integra Telecom SIP Solutions How to Configure the Avaya IP Office 6.1 for use with Integra Telecom SIP Solutions Overview This document provides a reference for configuration of the Avaya IP Office to connect to Integra Telecom SIP

More information

1) How do I setup my SIP trunk for inbound/outbound calling? We authenticate IP-PBX SIP Trunking traffic by:

1) How do I setup my SIP trunk for inbound/outbound calling? We authenticate IP-PBX SIP Trunking traffic by: 1) How do I setup my SIP trunk for inbound/outbound calling? We authenticate IP-PBX SIP Trunking traffic by: IP Authentication (IP address) or Digest Authentication (account and SIP password) After you

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

VOIP THE ULTIMATE GUIDE VERSION 1.0. 9/23/2014 onevoiceinc.com

VOIP THE ULTIMATE GUIDE VERSION 1.0. 9/23/2014 onevoiceinc.com VOIP THE ULTIMATE GUIDE VERSION 1.0 9/23/2014 onevoiceinc.com WHAT S IN THIS GUIDE? WHAT IS VOIP REQUIREMENTS OF A VOIP SYSTEM IMPLEMENTING A VOIP SYSTEM METHODS OF VOIP BENEFITS OF VOIP PROBLEMS OF VOIP

More information

Operation Manual Voice Overview (Voice Volume) Table of Contents

Operation Manual Voice Overview (Voice Volume) Table of Contents Operation Manual Voice Over (Voice Volume) Table of Contents Table of Contents Chapter 1 Voice Over... 1-1 1.1 Introduction to VoIP... 1-1 1.1.1 VoIP System... 1-1 1.1.2 Basic VoIP Call Flow... 1-2 1.1.3

More information

BroadCloud PBX Customer Minimum Requirements

BroadCloud PBX Customer Minimum Requirements BroadCloud PBX Customer Minimum Requirements Service Guide Version 2.0 1009 Pruitt Road The Woodlands, TX 77380 Tel +1 281.465.3320 WWW.BROADSOFT.COM BroadCloud PBX Customer Minimum Requirements Service

More information

FortiVoice. Version 7.00 VoIP Configuration Guide

FortiVoice. Version 7.00 VoIP Configuration Guide FortiVoice Version 7.00 VoIP Configuration Guide FortiVoice Version 7.00 VoIP Configuration Guide Revision 2 14 October 2011 Copyright 2011 Fortinet, Inc. All rights reserved. Contents and terms are subject

More information

1 SIP Carriers. 1.1.1 Warnings. 1.1.2 Vendor Contact Vendor Web Site : http://www.wind.it. 1.1.3 Versions Verified SIP Carrier status as of 9/11/2011

1 SIP Carriers. 1.1.1 Warnings. 1.1.2 Vendor Contact Vendor Web Site : http://www.wind.it. 1.1.3 Versions Verified SIP Carrier status as of 9/11/2011 1 SIP Carriers 1.1.1 Warnings Check the SIP 3 rd Party SIP Carrier Matrix for certification status, and supported features. More info about the SIP 3 rd Party SIP Carrier Matrix can be found in the SIP

More information

EXPLOITING SIMILARITIES BETWEEN SIP AND RAS: THE ROLE OF THE RAS PROVIDER IN INTERNET TELEPHONY. Nick Marly, Dominique Chantrain, Jurgen Hofkens

EXPLOITING SIMILARITIES BETWEEN SIP AND RAS: THE ROLE OF THE RAS PROVIDER IN INTERNET TELEPHONY. Nick Marly, Dominique Chantrain, Jurgen Hofkens Nick Marly, Dominique Chantrain, Jurgen Hofkens Alcatel Francis Wellesplein 1 B-2018 Antwerp Belgium Key Theme T3 Tel : (+32) 3 240 7767 Fax : (+32) 3 240 8485 E-mail : Nick.Marly@alcatel.be Tel : (+32)

More information

MITEL SIP CoE. Technical. Configuration Notes. Configure MCD 6.X for use with babytel SIP trunks. SIP CoE 13-4940-00266

MITEL SIP CoE. Technical. Configuration Notes. Configure MCD 6.X for use with babytel SIP trunks. SIP CoE 13-4940-00266 MITEL SIP CoE Technical Configuration Notes Configure MCD 6.X for use with babytel SIP trunks SIP CoE 13-4940-00266 NOTICE The information contained in this document is believed to be accurate in all respects

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

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

Table of Contents. Confidential and Proprietary

Table of Contents. Confidential and Proprietary Table of Contents About Toshiba Strata CIX and Broadvox SIP Trunking... 1 Requirements... 2 Purpose, Scope and Audience... 3 What is SIP Trunking?... 4 Business Advantages of SIP Trunking... 4 Technical

More information

Gateways and Their Roles

Gateways 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 information

Mediatrix 3000 with Asterisk June 22, 2011

Mediatrix 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 information

Digium Switchvox AA65 PBX Configuration

Digium Switchvox AA65 PBX Configuration Digium Switchvox SIP Trunking using Optimum Business SIP Trunk Adaptor and the Digium Switchvox AA65 IP-PBX v23695 Goal The purpose of this configuration guide is to describe the steps needed to configure

More information

An Introduction to VoIP Protocols

An Introduction to VoIP Protocols An Introduction to VoIP Protocols www.netqos.com Voice over IP (VoIP) offers the vision of a converged network carrying multiple types of traffic (voice, video, and data, to name a few). To carry out this

More information

5. DEPLOYMENT ISSUES Having described the fundamentals of VoIP and underlying IP infrastructure, let s address deployment issues.

5. DEPLOYMENT ISSUES Having described the fundamentals of VoIP and underlying IP infrastructure, let s address deployment issues. 5. DEPLOYMENT ISSUES Having described the fundamentals of VoIP and underlying IP infrastructure, let s address deployment issues. 5.1 LEGACY INTEGRATION In most cases, enterprises own legacy PBX systems,

More information

VoIP Network Configuration Guide

VoIP Network Configuration Guide The owner friendly phone system for small business VoIP Network Configuration Guide Release 7.10 Copyright 2011 Fortinet, Inc. All rights reserved. Fortinet, FortiGate, FortiGuard, FortiCare, FortiManager,

More information

Packetized Telephony Networks

Packetized 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 information

PETER CUTLER SCOTT PAGE. November 15, 2011

PETER 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 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

Note: As of Feb 25, 2010 Priority Telecom has not completed FXS verification of fax capabilities. This will be updated as soon as verified.

Note: As of Feb 25, 2010 Priority Telecom has not completed FXS verification of fax capabilities. This will be updated as soon as verified. 1 SIP Carriers 1.1 Priority Telecom 1.1.1 Warnings Check the SIP 3 rd Party SIP Carrier Matrix for certification status, and supported features. More info about the SIP 3 rd Party SIP Carrier Matrix can

More information

VoIP Solutions Guide Everything You Need to Know

VoIP Solutions Guide Everything You Need to Know VoIP Solutions Guide Everything You Need to Know Simplify, Save, Scale VoIP: The Next Generation Phone Service Ready to Adopt VoIP? 10 Things You Need to Know 1. What are my phone system options? Simplify,

More information

SIP Trunking Guide: Get More For Your Money 07/17/2014 WHITE PAPER

SIP Trunking Guide: Get More For Your Money 07/17/2014 WHITE PAPER SIP Trunking Guide: Get More For Your Money 07/17/2014 WHITE PAPER Overview SIP trunking is the most affordable and flexible way to connect an IP PBX to the Public Switched Telephone Network (PSTN). SIP

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

The Basics. Configuring Campus Switches to Support Voice

The Basics. Configuring Campus Switches to Support Voice Configuring Campus Switches to Support Voice BCMSN Module 7 1 The Basics VoIP is a technology that digitizes sound, divides that sound into packets, and transmits those packets over an IP network. VoIP

More information

How to Configure the Cisco UC500 for use with Integra Telecom SIP Solutions

How to Configure the Cisco UC500 for use with Integra Telecom SIP Solutions How to Configure the Cisco UC500 for use with Integra Telecom SIP Solutions Overview: This document provides a reference for configuration of the Cisco UC500 IP PBX to connect to Integra Telecom SIP Trunks.

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

Allstream Converged IP Telephony

Allstream 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 information

Internet Technology Voice over IP

Internet Technology Voice over IP Internet Technology Voice over IP Peter Gradwell BT Advert from 1980s Page 2 http://www.youtube.com/v/o0h65_pag04 Welcome to Gradwell Gradwell provides technology for every line on your business card Every

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

Integrating VoIP Phones and IP PBX s with VidyoGateway

Integrating 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 information

Auto Attendants. Call Management

Auto Attendants. Call Management Auto Attendants Customer Portal Top Level Auto Attendant (Always On) Multiple Top Level Auto Attendants (Always on) Top Level Auto Attendant (Time Based) Sub-Level Auto Attendants Web based user interface

More information

SIP Trunking Service Configuration Guide for MegaPath

SIP Trunking Service Configuration Guide for MegaPath Notice Note that when converting this document from its original format to a.pdf file, some minor font and format changes may occur. When viewing and printing this document, we cannot guarantee that your

More information

icall VoIP (User Agent) Configuration

icall VoIP (User Agent) Configuration icall VoIP (User Agent) Configuration 1 General 1.1 Topic General Document summarizing the general requirements for the configuration of VoIP hardware and / or software to utilize the icall service. 1.2

More information

SIP Trunking DEEP DIVE: The Service Provider

SIP 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 information

Voice over IP (VoIP) for Telephony. Advantages of VoIP Migration for SMBs BLACK BOX. 724-746-5500 blackbox.com

Voice over IP (VoIP) for Telephony. Advantages of VoIP Migration for SMBs BLACK BOX. 724-746-5500 blackbox.com Voice over IP (VoIP) for Telephony Advantages of VoIP Migration for SMBs BLACK BOX Hybrid PBX VoIP Gateways SIP Phones Headsets 724-746-5500 blackbox.com Table of Contents Introduction...3 About Voice

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

Configuration guide for Switchvox and Cbeyond.

Configuration guide for Switchvox and Cbeyond. Configuration guide for Switchvox and Cbeyond. This document will guide a Switchvox administrator through configuring the system to utilize Cbeyond s BeyondVoice with SIPconnect service. After you have

More information

IP Telephony Basics. Part of The Technology Overview Series for Small and Medium Businesses

IP Telephony Basics. Part of The Technology Overview Series for Small and Medium Businesses IP Telephony Basics Part of The Technology Overview Series for Small and Medium Businesses What is IP Telephony? IP Telephony uses the Internet Protocol (IP) to transmit voice or FAX traffic over a public

More information

Technical Configuration Notes

Technical Configuration Notes MITEL SIP CoE Technical Configuration Notes Configure MCD for use with OpenIP SIP Trunking service SIP CoE 11-4940-00186 NOTICE The information contained in this document is believed to be accurate in

More information

Configuration of Applied VoIP Sip Trunks with the Toshiba CIX40, 100, 200 and 670

Configuration of Applied VoIP Sip Trunks with the Toshiba CIX40, 100, 200 and 670 Configuration of Applied VoIP Sip Trunks with the Toshiba CIX40, 100, 200 and 670 Businesses Save Money with Toshiba s New SIP Trunking Feature Unlike gateway based solutions, Toshiba s MIPU/ GIPU8 card

More information

VoIP / SIP Planning and Disclosure

VoIP / SIP Planning and Disclosure VoIP / SIP Planning and Disclosure Voice over internet protocol (VoIP) and session initiation protocol (SIP) technologies are the telecommunication industry s leading commodity due to its cost savings

More information

Hosted VoIP Feature Set

Hosted VoIP Feature Set Hosted VoIP Set AUTO ATTENDANTS Customer Portal Top Level Auto Attendant (Always On) Multiple Top Level Auto Attendants (Always on) Top Level Auto Attendant (Time Based) Sub-Level Auto Attendants Web based

More information

Is Your Network Ready for VoIP? > White Paper

Is Your Network Ready for VoIP? > White Paper > White Paper Tough Questions, Honest Answers For many years, voice over IP (VoIP) has held the promise of enabling the next generation of voice communications within the enterprise. Unfortunately, its

More information

VOIP Security Essentials. Jeff Waldron

VOIP Security Essentials. Jeff Waldron VOIP Security Essentials Jeff Waldron Traditional PSTN PSTN (Public Switched Telephone Network) has been maintained as a closed network, where access is limited to carriers and service providers. Entry

More information

Guideline for SIP Trunk Setup

Guideline for SIP Trunk Setup Guideline for SIP Trunk Setup with ZONETEL Table of contents Sample sip.conf (it applies to asterisk 1.4.x)...3 Sample elastix setup... 3 Ports required... 4 Caller ID...4 FAQ... 5 After i dial out, the

More information

1.1.3 Versions Verified SIP Carrier status as of 18 Sep 2014 : validated on CIC 4.0 SU6.

1.1.3 Versions Verified SIP Carrier status as of 18 Sep 2014 : validated on CIC 4.0 SU6. 1 SIP Carriers 1.1 Telstra 1.1.1 Warnings Check the SIP 3 rd Party SIP Carrier Matrix for certification status, and supported features. More info about the SIP 3 rd Party SIP Carrier Matrix can be found

More information

AUTO ATTENDANTS CALL MANAGEMENT

AUTO ATTENDANTS CALL MANAGEMENT AUTO ATTENDANTS Customer Portal Top Level Auto Attendant (Always On) Multiple Top Level Auto Attendants (Always on) Top Level Auto Attendant (Time Based) Sub-Level Auto Attendants Web based user interface

More information

Hosted Business Class VoIP Features

Hosted Business Class VoIP Features Hosted Business Class VoIP s Customer Portal Web based user interface that allows users to configure their PBX, create call queues and groups, view call detail records and billing information, listen to

More information

Version 0.1 June 2010. Xerox WorkCentre 7120 Fax over Internet Protocol (FoIP)

Version 0.1 June 2010. Xerox WorkCentre 7120 Fax over Internet Protocol (FoIP) Version 0.1 June 2010 Xerox WorkCentre 7120 Fax over Internet Protocol (FoIP) Thank you for choosing the Xerox WorkCentre 7120. Table of Contents Introduction.........................................

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

SIP Trunk Configuration V/IPedge Feature Description 5/22/13

SIP Trunk Configuration V/IPedge Feature Description 5/22/13 SIP Trunk Configuration V/IPedge Feature Description 5/22/13 OVERVIEW Session Initiation Protocol (SIP) is an application layer protocol used for establishing sessions in an IP network. SIP trunks allow

More information

Getting Started. 16-Channel VoIP Gateway Card. Model No. KX-TDA0490

Getting Started. 16-Channel VoIP Gateway Card. Model No. KX-TDA0490 16-Channel VoIP Gateway Card Getting Started Model No. KX-TDA0490 Thank you for purchasing a Panasonic 16-Channel VoIP Gateway Card. Please read this manual carefully before using this product and save

More information