ACCELERATOR 6.3 TDM PBX INTEGRATION GUIDE
|
|
|
- Elijah Owens
- 10 years ago
- Views:
Transcription
1 ACCELERATOR 6.3 TDM PBX INTEGRATION GUIDE April 2014 Tango Networks, Inc. phone: Parkwood Blvd, Suite 500 fax: Frisco, Texas USA
2 Tango Networks, Inc. This software is protected by copyright law and international treaties, and is the confidential and proprietary information of Tango Networks, Inc. Unauthorized reproduction, use, or distribution of this software, or any portion of it, may result in severe civil and criminal penalties, and will be prosecuted to the maximum extent possible under the law. The software described in this document is furnished under license agreement and may only be used in accordance with the terms of the agreement. Tango Networks and Abrazo are trademarks of Tango Networks, Inc. All other trademarks used herein are the property of their respective owners and are used for identification purposes only Tango Networks, Inc. Tango Networks, Abrazo and E=fmc2 are trademarks or registered trademarks of Tango Networks, Inc. All other trademarks or service marks are the property of their respective owners. Specifications and features are subject to change without notice. April 2014 CONFIDENTIAL Page 2 of 16
3 TABLE OF CONTENTS INTRODUCTION... 4 SUPPORTED VERSIONS... 4 SIP TO PRI CONVERSION... 4 SIMULTANEOUS RINGING (SIMRING)... 4 TDM PBX DEPENDENCIES... 4 License Key... 4 Trunking Interfaces... 4 TDM PBX INTEGRATION REQUIREMENTS... 5 INSTALLATION OF SIP/PRI GATEWAY... 5 TRANSLATIONS PROVISIONING... 5 CONFIGURATION OF SIMRING FEATURE... 5 ACCELERATOR PROVISIONING... 6 Voice Network : PBX/Trunk Dial Plan... 6 Voice Network : Extension Ranges... 7 Voice Network : Carrier Gateways... 7 Voice Network: Voice Mail... 8 Subscriber Dial Plan/Subscriber... 8 FEATURE INTERACTIONS... 9 ACCELERATOR PBX LEVEL 1 INTEGRATION... 9 ACCELERATOR PBX LEVEL 2 INTEGRATION...12 ACCELERATOR PBX LEVEL 3 INTEGRATION...13 APPENDIX A: MULTIPLE CALL APPEARANCE FOR NORTEL DMS-100 PBX ACRONYMS April 2014 CONFIDENTIAL Page 3 of 16
4 Introduction The Accelerator extends enterprise PBX functionality to mobile devices allowing end users to be more productive and accessible when out of the office. The Accelerator integrates mobile devices with existing Private Branch Exchanges (PBXs) so that the PBX sees the mobile device as simply another desk phone. This allows the existing PBX feature set to be applied consistently across both devices. Mobile specific functionality is then layered on top. Supported Versions This document addresses the required configuration changes on TDM PBXs (PBXs that do not support SIP trunk or SIP line interfaces) to enable integrated operation with the Accelerator. SIP to PRI conversion The Accelerator uses the SIP protocol for call control of the calls made and received by Accelerator subscribers. A SIP/PRI gateway is therefore used to provide the necessary inter-working function for these releases. This requires the provisioning of PRI trunks on the TDM PBX along with translations and routing as described in section TDM PBX Integration Requirements. Simultaneous Ringing (SIMRING) If the Accelrator mobile is to be used in conjunction with a desk phone (or soft client), the TDM PBX must support a SIMRING feature that allows an incoming call to ring one or more devices simultaneously. TDM PBX Dependencies License Key To use the Accelerator with TDM PBXs, the following software licenses may be required on the PBX: One SIMRING license per Accelerator subscriber who wishes to use both their desk phone (or soft client) and their Accelerator mobile Trunking Interfaces To use the Accelerator with TDM PBXs, PRI trunks are required on the TDM PBX to interwork with the SIP/PRI gateway. April 2014 CONFIDENTIAL Page 4 of 16
5 TDM PBX Integration Requirements This section describes the requirements for the Accelerator to integrate with a TDM PBX. The following checklist itemizes the items required for integration: SIP/PRI Gateway (e.g. Audiocodes Mediant line of gateways) Available PRI trunk ports on TDM PBX SIMRING feature enabled on TDM PBX Provisioning of translations capability Pilot Numbers and Accelerator alias numbers Provisioning of PBX and subscribers on Accelerator Enterprise solution Installation of SIP/PRI gateway The SIP/PRI gateway needs to be installed in a location accessible by the PRI spans from the TDM PBX and by an Ethernet connection to a network segment routable to the Accelerator Enterprise servers. The PRI spans are connected between the TDM PBX trunk peripheral and the SIP/PRI gateway. Tango Networks has completed interoperability testing with Mediant AudioCodes 2000 gateway, net VX gateway, and AdTran gateway. Please check with Tango for additional gateway interop results. Translations Provisioning Pilot numbers need to be allocated to route calls into the enterprise for handling by Accelerator. Translations need to be set up to route these inbound calls to the pilot numbers over the trunk that is connected to the PRI gateway and then to the Accelerator. Alias numbers also need to be set up to support the simultaneous ringing capability of the desk phone. These alias numbers should be set up to route over the trunk to the Accelerator. Configuration of SIMRING feature The SIMRING feature allows an incoming call to ring a group of directory numbers simultaneously. To support the Accelerator, the SIMRING feature is configured for each Accelerator subscriber s desk phone to enable simultaneous ringing of the user s desk phone and an Accelerator alias. April 2014 CONFIDENTIAL Page 5 of 16
6 Accelerator Provisioning Note: This document assumes that the Accelerator has already been provisioned with: - Enterprise information - Wireless carrier information The integration process includes the following steps: 1. Define a Trunk Dial Plan for the TDM PBX in the Accelerator (required). 2. Define the TDM PBX in the Accelerator (required). Define a SIP Trunk Group/Trunk to route traffic from the TDM PBX to the Accelerator (required). Define Pilot Numbers (optional but recommended) - PSTN routable number(s) that represent the mobile route into the enterprise network. The enterprise network must be provisioned to route calls for each pilot number to the Accelerator. Define the Call Service Pilot Numbers (required if using the Call Move service)- Used for the Call Move service. Define Least Cost Route (optional) information for SIP trunk traffic from the Accelerator to the TDM PBX (optional). 3. Define the Enterprise Ranges used in the enterprise. 4. Define the Carrier Gateways (required if your enterprise routes calls between the wireless carrier and the Accelerator). 5. Define the Voice Mail system used with the TDM PBX (recommended). 6. Define Subscriber Dial Plans in the Accelerator (required). 7. Define Accelerator subscribers that use the TDM PBX (required). The steps below describe the unique configuration areas needed to integrate the TDM with the Accelerator. Refer to the Accelerator Provisioning Guide for a comprehensive explanation of Accelerator provisioning. Voice Network : PBX/Trunk Dial Plan 1. Add Trunk Dial Plan (required) There are no unique configuration items for Trunk Dial Plans and the TDM PBX. Refer to the Accelerator Provisioning Guide, Voice Networks, PBXs, Add Trunk Dial Plan section. 2. Add PBX (required) Refer to the Accelerator Provisioning Guide, Voice Networks, PBXs, Add New PBX section. Add Trunk Group/Trunk (required) o A SIP trunk is created to route traffic from the Accelerator to SIP PRI Gateway associated with the TDM PBX. Multiple trunks to different gateways may be created for scalability/redundancy reasons. The April 2014 CONFIDENTIAL Page 6 of 16
7 o o Accelerator will randomly select trunks in a trunk group on a per-call basis and will route advance among trunks in the trunk group if a trunk is unavailable. When provisioning Trunk Groups/Trunks in the Accelerator. Trunk Group Request URI parameters are not used by the TDM PBX and therefore do not need to be provisioned. Refer to the Accelerator Provisioning Guide, Voice Networks, PBXs (Add/Modify), Add Trunk Groups/Trunk section. Add Pilot Numbers (required if your enterprise intends to route Pilot DNs through the TDM PBX) - No unique configuration areas required for provisioning Pilot Numbers to the Accelerator. Refer to the Accelerator Provisioning Guide, Voice Networks, PBXs (Add/Modify) section. Add Call Service Pilot Number (required if using the Call Move Service). No unique configuration items for Call Service Pilot Numbers. Refer to the Accelerator Provisioning Guide, Voice Networks, PBXs (Add/Modify) section. Add Least Cost Routing (optional) in the Accelerator for the TDM PBX. No unique configuration items for Least Cost Routing. Refer to the Accelerator Provisioning Guide, Voice Networks, Least Cost Routes, Add Least Cost Routes section. Voice Network : Extension Ranges 3. Add Extension Ranges (required) No unique configuration items for adding an Extension Range. Refer to the Accelerator Provisioning Guide, Voice Networks, Extension Ranges, Add New Extension Ranges section. Voice Network : Carrier Gateways 4. Carrier Gateways (required if your enterprise routes calls between the wireless carrier and the Accelerator) If your enterprise intends to use an SBC or Carrier Gateway between the carrier and the enterprise, then add a Carrier Gateway. Be sure to supply the Outbound Domain as required by the add Carrier Gateway page. If you are using Pilot numbers to route calls from the wireless carrier to the Accelerator using the Carrier Gateway, provision the Pilot Numbers on the Carrier Gateway s main page. Depending on your configuration, the Pilot numbers here may need to be E.164 routable. Add Trunk Groups/Trunk (required if using a Carrier Gateway) o o Host Address, Port and Transport Type should match the value of the carrier s SIP endpoint. If the hostname is entered, the Accelerator queries the DNS server for the associated DNS SRV records. This allows for redundancy and/or load balancing based on the DNS SRV records if the customer has deployed multiple TDM servers. Refer to the Accelerator Provisioning Guide, Voice Networks, Carrier Gateway (Add/Modify), Add Trunk Groups/Trunks section. Trunk Group Request URI parameters are normally not required by most SIP endpoints, however, consult your carrier s instructions. Add Least Cost Routing (optional) - No unique configuration items for Least Cost Routing. Refer to the Accelerator Provisioning Guide, Voice Networks, Carrier Gateway, Least Cost Routes, Add Least Cost Routes section. April 2014 CONFIDENTIAL Page 7 of 16
8 Voice Network: Voice Mail 3. Add Voice Mail (recommended) for the TDM PBX in the Accelerator. Select either SIP or SMDI as the Voice Mail Server Type. Refer to the Accelerator Provisioning Guide, Voice Networks, Voice Mail Servers, Add Voice Mail Server section. Note: For Callpilot voic , select SMDI as Voice Mail Server Type. Subscriber Dial Plan/Subscriber 4. Add Subscriber Dial Plan - (required) - No unique configuration items for adding Subscriber Dial Plans. Refer to the Accelerator Provisioning Guide, Subscribers, Add Subscriber Dial Plan section. 5. Add Subscribers (required) - No unique configuration items for adding Subscribers. Refer to the Accelerator Provisioning Guide, Subscribers, Add Subscriber section. April 2014 CONFIDENTIAL Page 8 of 16
9 Feature Interactions This section explains the PBX features that have been integrated with this Accelerator release. Accelerator PBX Level 1 Integration Table 1 Accelerator PBX Level 1 Integration Feature Support Comments Abbreviated Dialing Ad Hoc Conferencing (Internal to PBX) No Ad Hoc Conferencing (External to PBX) Uses an external IP Media 2000 conference server. Refer to the Accelerator Provisioning Guide, Voice Networks, Conference Servers section. Call Forward All (Desk) Call Forward All Activation from Mobile No Call Forward Busy (Desk) Call Forward Busy Activation from Mobile No Call Forward No Answer (Desk) Call Forward No Answer Activation from Mobile No Call Hold and Retrieve (Mobile) Call Line Identification (CLID) Call Transfer Blind Call Transfer Consultative Call Waiting and Retrieve Direct Inward Dialing Direct Outward Dialing Directory Dial Flexible Dialing Support Intelligent Call Delivery Least Cost Routing Meet-Me Conference Multiple Calls per Line PBX Do Not Disturb (Desk) PBX Do Not Disturb (Mobile) Single Number Services Voice Mail Message Waiting Indication Integration Level 1 provides the subscriber basic services that are commonly used. The following features are considered Level 1: Abbreviated Dialing - Allows extension dialing or internal dialing from the desktop phone. The Accelerator allows the user to dial these same abbreviated numbers from the mobile phone. Ad Hoc Conferencing (Internal to PBX)- Allows an Accelerator subscriber to initiate a reservation-less conference from a mobile phone using the conference resources of the PBX. Ad Hoc Conferencing using (External to PBX) - Allows an Accelerator subscriber to initiate a reservation-less conference from a mobile phone using external media server located in the enterprise. April 2014 CONFIDENTIAL Page 9 of 16
10 Call Forward All (Desk) - Allows users to forward all calls to another destination including those calls to the mobile number. This feature is activated via the desk phone. Call Forward Activation on Mobile - Allows users to forward all calls to another destination. Users enter a feature access code on their mobile phone to activate or deactivate call forwarding. Call Forward Busy (Desk) - Allows users to forward calls (including those to their mobile number) to another destination when their device is busy. Users activate or deactivate the Call Forward Busy capability with a Call Forward Busy feature button from their desk phone Call Forward Busy Activation on the Mobile - Allows users to forward calls to another destination when their device is busy. Users activate or deactivate the Call Forward Busy capability with a feature access code from their mobile phone. Call Forward No Answer (Desk) - Allows users to forward calls (including calls to their mobile number) to another destination when their device is not answered. Users activate or deactivate the Call Forward No Answer capability with a Call Forward No Answer feature button from their desk phone Call Forward No Answer Activation on Mobile - Allows users to forward calls to another destination when their device is not answered. Users activate or deactivate the Call Forward No Answer capability with a feature access code from their mobile phone. Call Hold and Retrieve (Mobile) Lets users temporarily disconnect from a call, use the telephone for another call, and then return to the original call. The Accelerator supports this capability in concert with the wireless network. Call Line Identification (CLID) Provides the user information about the calling party. The Accelerator supports calling line identification when it is the called party. The Accelerator also supports ensuring that the enterprise identity of the caller is preserved when a call is initiated from the mobile phone. In this case although the call is made from a mobile, the calling line ID will be that of the Accelerator user's desktop phone. The enterprise main number may also optionally be used in place of a subscriber s DID for off-net calling. Call Transfer Lets users move a currently established call from their mobile phone to another destination. This is implemented by the user entering a mid-call feature code followed by the transfer to number. There are two types of call transfers that are supported by this functionality: o o Blind Call Transfer Call is transferred without interaction between the user who initiated the transfer and the transfer destination. Consultative Call Transfer - Call is transferred allowing interaction between the user who initiated the transfer and the transfer destination. Call Waiting and Retrieve - Provides users with an audible alert in the voice stream that a new incoming call is waiting. The user can retrieve the call from the desk phone. The Accelerator supports call waiting in concert with the wireless network. Call Waiting tones are provided by the mobile phone when an incoming call is waiting, and waiting calls can be retrieved from the mobile phone. April 2014 CONFIDENTIAL Page 10 of 16
11 Direct Inward Dialing - Allows the desk phone to be directly accessed from the PSTN. The Accelerator supports enterprise Direct Inward Dialing. Direct Outward Dialing - Allows users inside an enterprise to dial directly to an external number. The Accelerator supports the mobile device dialing directly to an external number. Directory Dial - Lets users select numbers to dial from a corporate or personal directory. The Accelerator supports using a personal directory on the phone and handles the translations of those digits into on-net network numbers if appropriate. In addition, the Accelerator support a corporate directory look up capability for access to the corporate address book. Flexible Dialing Support - The Accelerator has a flexible dialing plan enabling PBX services to be provided to mobile users. Intelligent Call Delivery - Ensures that both the desk phone and mobile phone ring when the dialed number is an Accelerator subscriber. Least Cost Routing For mobile originations and terminations, the Accelerator ensures that the least cost route is used. This results in the enterprise voice network being used to route the call as much as possible, reducing PSTN interconnect costs, and other voice costs such as roaming. Meet-Me Conference - Allows users to set up a dial-in conference of up to six parties. The mobile user can participate in the meet-me conference by dialing the conference bridge. Multiple Calls per Line Allows multiple calls to be delivered to a single number and have the incoming call information displayed to the user. The Accelerator supports this feature on the mobile phone based on the ability to support call waiting for mobile phone devices. Mobile devices typically show a maximum of two lines per mobile phone. PBX Do Not Disturb (Desk) Allows users to activate or deactivate the Do Not Disturb capability by pressing a button or a softkey from their desk phone. When active, Do Not Disturb will not ring the mobile or desk phone. PBX Do Not Disturb (Mobile) - Allows users to activate or deactivate the Do Not Disturb capability by entering a feature access code from their mobile phone. When active, Do Not Disturb will not ring the mobile phone however the desk phone will continue to ring. Single Number Services - A single phone number that a subscriber publishes to communicate with others. When this single number is dialed, the subscriber s enterprise desktop phone as well as mobile phone will ring. Voice Mail Waiting Indication - Provides a visible indication on the mobile phone that there is a message waiting in the voice mail system. The Accelerator supports supplying a Message Waiting indication on the mobile phone that indicates that there are voice mail messages in the enterprise voice mail system. April 2014 CONFIDENTIAL Page 11 of 16
12 Accelerator PBX Level 2 Integration Table 2 Accelerator PBX Level 2 Integration Feature Support Comments Call Accounting Codes No Call Coverage/Hunt Groups Call Line Identification Restriction (CLIR) (Mobile) Call Pull (Desk->Mobile Call Move) No Call Push (Mobile->Desk Call Move) Class of Restriction (COR) (PBX) No Class of Service (COS) (PBX) No Integration Level 2 provides the subscriber more advanced features than more commonly used basic features. The following features are considered Level 2 integration targets: Call Accounting Codes - To support the mobile office environment, client billing must be supported when the user is away from his/her desktop phone and using a mobile phone. For example, law offices, accounting firms, consulting firms and other organizations benefit from tracking the length of a call for a client. Client billing is achieved by having the user enter a code to specify that the call relates to a specific client matter or account. The code, which is often referred to as a client matter code (CMC) or call accounting code, can be assigned to customers, students, or other populations for call accounting and billing purposes. Call accounting codes are used by enterprise to manage call accounting. Call Coverage/Hunt Groups - Allows a group of extensions to be set up to handle multiple calls to a single telephone number. For each call to the number, the PBX hunts for an available extension in the hunt group and connects the call to that extension. The Accelerator can be defined as one of the extensions. Call Line Identification Restriction (CLIR) (Mobile) - Allows the user to restrict their calling line information from being displayed to the called number. The Accelerator supports restriction of calling line identification from mobile phones. Enterprise identity will be replaced; however restriction code will be preserved. Call Pull (Desk->Mobile Call Move)- Allows a subscriber to move a phone call between the desk phone and the mobile phone. Feature is invoked from the mobile phone. Call Push (Mobile->Desk Call Move)- Allows a subscriber to move a phone call between the desk phone and the mobile phone. Feature is be invoked from the mobile phone. Class of Restriction (COR) (PBX) - Defines the restrictions that apply when a user places or receives a call. The Accelerator supports COR for mobile originated calls. Class of Service (COS) (PBX) - Allows or denies user access to some system features. The Accelerator supports COS for mobile originated calls over SIP lines. April 2014 CONFIDENTIAL Page 12 of 16
13 Accelerator PBX Level 3 Integration Integration Level 3 provides features that are specific to the PBX or specific to vertical markets. The Accelerator does not support any Level 3 integration with the TDM PBX. April 2014 CONFIDENTIAL Page 13 of 16
14 Appendix A: Multiple Call Appearance for Nortel DMS-100 PBX For a Nortel DMS-100 PBX, the desk phones can be configured with Multiple Appearance Directory Number (MADN). The MADN allows each DN to have more than one line. Currently on some older loads of the DMS-100, desks with MADN Multiple Call Appearance (MCA) cannot be datafilled with Simring. We recommend the use of Key Short Hunting (KSH), which is compatible with Simring, and would act as if the desk has two lines for the one DN. The following steps should be taken to convert from MCA to KSH with Simring: 6. Remove MCA from the DN. 7. Assign a new fake DN to the 2 nd line of the desk phone. 8. Add Simring to the first line. 9. Use KSH for both lines. 10. Remove CFB and CFD voic re-direct from the first line. 11. KSH Default should route to a voic number, in the event that both lines are busy. 12. Add CFD voic re-direct to the 1 st and 2 nd line. 13. Add Number Replacement on the 2 nd line to represent the 1 st DN. This would allow the voic system to recognize the correct voice mailbox. The only restriction on using the above method is that if the first line is busy, Simring does not activate. This results in the 2 nd desk line ringing, but the mobile would not. April 2014 CONFIDENTIAL Page 14 of 16
15 Acronyms Table 3 Term Accelerator CA CDR CFA CFB CFNA CLI CLID CLIR COR COS CTI DID DN DTMF IPDR Mobilizer MWI NAT PBX PDN PSTN SIM Ring SIP SMDI SOAP TDM TLDN TLS Acronyms Definition Tango Enterprise Certificate Authority Call Detail Record Call Forward All Calls Call Forward Busy Call Forward Not Answered Command Line Interface Calling Line Identification Calling Line Identification Restriction Class of Restriction Class of Service Computer Telephony Integration Direct Inward Dial Directory Number Dual-Tone Multi-Frequency Internet Protocol Data Record Tango Carrier Message Waiting Indication Network Address Translation Private Branch Exchange Pilot Directory Number Public Switched Telephone Network Simultaneous Ring Session Initiation Protocol Simplified Message Desk Interface Simple Object Access Protocol Time Division Multiplex Temporary Location Directory Number Transport Layer Security April 2014 CONFIDENTIAL Page 15 of 16
16 Tango Networks, Inc Parkwood Blvd, Suite 500 Frisco, Texas USA phone: fax:
ACCELERATOR 6.3 SIP TRUNK PBX INTEGRATION GUIDE
ACCELERATOR 6.3 SIP TRUNK PBX INTEGRATION GUIDE April 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
ACCELERATOR 6.3 ASTERISK LINES INTEGRATION GUIDE
ACCELERATOR 6.3 ASTERISK LINES INTEGRATION GUIDE January 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
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
ACCELERATOR 6.3 AASTRA MX-ONE INTEGRATION GUIDE
ACCELERATOR 6.3 AASTRA MX-ONE INTEGRATION GUIDE September 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
ACCELERATOR 6.3 MITEL 3300 INTEGRATION GUIDE
ACCELERATOR 6.3 MITEL 3300 INTEGRATION GUIDE SINGLE NODE ACCELERATOR July 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
ACCELERATOR 6.3 ALARMS AND OPERATIONAL MEASUREMENTS
ACCELERATOR 6.3 ALARMS AND OPERATIONAL MEASUREMENTS 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
ACCELERATOR 6.4 CISCO UNITY 3.1/4.1 INTEGRATION GUIDE
ACCELERATOR 6.4 CISCO UNITY 3.1/4.1 INTEGRATION GUIDE March 2015 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
BROADSOFT PARTNER CONFIGURATION GUIDE VEGASTREAM VEGA 100
BROADSOFT PARTNER CONFIGURATION GUIDE VEGASTREAM VEGA 100 JULY 2005 Version 1.0 BroadWorks Guide Copyright Notice Copyright 2005 BroadSoft, Inc. All rights reserved. Any technical documentation that is
Administration. Avaya Business Communications Manager Find Me/Follow Me. Introduction. Find Me/Follow Me Fundamentals
Avaya Business Communications Manager Find Me/Follow Me Administration Introduction Find Me/Follow Me allows simultaneous ringing on up to five different external destinations. Users and administrators
Quick Setup Guide. Integration of Aastra MX-ONE / Aastra 700 and Microsoft Lync Server 2010
Quick Setup Guide Integration of Aastra MX-ONE / Aastra 700 and Microsoft Lync Server 2010 Aastra MX-ONE TM and Aastra 700 running Telephony Server software V.4.1 SP3 and later Aastra Telecom Sweden AB
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
Publication Information This document is a publication of IPVision S.A. 112 Bernardo de Irigoyen, 4th Floor (C1072AAD) Buenos Aires, Argentina
IP Centrex Data Sheet The most effective turn-key hosted virtual PBX solution for Clarent Networks Publication Information This document is a publication of IPVision S.A. 112 Bernardo de Irigoyen, 4th
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
Businesses Save Money with Toshiba s New SIP Trunking Feature
TOSHIBA Strata CIX Product Bulletin PBCIX-0056 Dec. 7, 2007 Businesses Save Money with Toshiba s New SIP Trunking Feature For business trying to save money on telecommunications tariffs, conventional technology
SIP Trunking Configuration with
SIP Trunking Configuration with Microsoft Office Communication Server 2007 R2 A Dell Technical White Paper End-to-End Solutions Team Dell Product Group - Enterprise THIS WHITE PAPER IS FOR INFORMATIONAL
: AudioCodes. Updated Since : 2007-05-28 READ THIS BEFORE YOU PROCEED
Microsoft Exchange Server 2007 Unified Messaging PBX Configuration Note: Inter-Tel 5000 with the Mediant 2000 Gateway using T1-CAS (In-band DTMF) By : AudioCodes Updated Since : 2007-05-28 READ THIS BEFORE
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
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.
Using 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
CLEARSPAN 911/E911 Overview
CLEARSPAN 911/E911 Overview Revision 09012014-1 Proprietary Notice This document contains sensitive and proprietary information and company trade secrets that are critical to Aastra business. This information
SIP 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
Updated Since : 2007-02-09
Microsoft Exchange Server 2007 Unified Messaging PBX Configuration Note: Avaya S8300 with AudioCodes Mediant 2000 using T1 CAS (In-band DTMF Tones) By : AudioCodes Updated Since : 2007-02-09 READ THIS
Management Summary for Unified Communications IP PBX
Management Summary for Unified Communications IP PBX Prepared By for YOU of General: The Unified Communication Internet Protocol Private Branch Exchange (UCIPPBX) is a fully realised 3 rd generation office
Motorola Solutions Integration Guide TEAM WSM to Avaya Session Manager with Nortel CS1000 Trunk Side Integration
Motorola Solutions Integration Guide TEAM WSM to Avaya Session Manager with Nortel CS1000 Trunk Side Integration 1 Integration Guide TEAM WSM to Avaya Session Manager and CS1000 Table of Contents 1.0 Summary...
TXI Telephony Product Book 2012 2012
TXI Telephony Product Book 2012 2012 PRICING, HOW TO This section details the common pricing scenarios for the following: Receptionist Call Centre SIP Trunking Fax to Email Messaging Receptionist: Receptionist
Feature and Technical
BlackBerry Mobile Voice System for SIP Gateways and the Avaya Aura Session Manager Version: 5.3 Feature and Technical Overview Published: 2013-06-19 SWD-20130619135120555 Contents 1 Overview...4 2 Features...5
IP Office 7.0 and BCM 6.0 SIP Interoperability Configuration Notes
IP Office 7.0 and BCM 6.0 SIP Interoperability Configuration Notes Abstract: This document provides information on how to configure a network solution with IP Office 7.0 and BCM 6.0 using SIP trunks. 2011
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
Cisco Unified Communications Manager (CUCM)
Spectralink 87-Series Wireless Telephone Cisco Unified Communications Manager (CUCM) Interoperability Guide 721-0016-000 Rev: B March 2015 Copyright Notice 2014-2015 Spectralink Corporation All rights
Application Notes for Configuring Avaya IP Office 9.0 with HIPCOM SIP Trunk Issue 1.0
Avaya Solution & Interoperability Test Lab Application Notes for Configuring Avaya IP Office 9.0 with HIPCOM SIP Trunk Issue 1.0 Abstract These Application Notes describe the procedures for configuring
Succession Multimedia Communications Portfolio MCP Interworking Basics NN10033-111. Standard MCP 1.1 FP1 (02.02) April 2003
Succession Multimedia Communications Portfolio MCP Interworking Basics NN10033-111 Standard MCP 1.1 FP1 (02.02) April 2003 Nortel Networks Confidential Overview 3 How is this chapter organized This chapter
Technical Configuration Notes
MITEL SIPCoE Technical Configuration Notes Configure Inn-Phone SIP Phone for use with MCD SIP CoE NOTICE The information contained in this document is believed to be accurate in all respects but is not
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
BroadSoft Partner Configuration Guide
BroadSoft Partner Configuration Guide Microsoft Lync 2010 SIP Trunking August 2012 Document Version 1.6 9737 Washingtonian Blvd Suite 350 Gaithersburg, MD USA 20878 Tel +1 301.977.9440 WWW.BROADSOFT.COM
mobile unified communications client and docking station
FREQUENTLY ASKED QUESTIONS mobile unified communications client and docking station What are the target customer characteristics of a Mobile UC subscriber? + Verizon Wireless as mobile carrier. Mobile
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
Evolution PBX User Guide for SIP Generic Devices
Evolution PBX User Guide for SIP Generic Devices Table of contents Introduction... 1 Voicemail... Using Voicemail... Voicemail Menu... Voicemail to Email... 3 Voicemail Web Interface... 4 Find Me Rules...
BroadSoft Partner Configuration Guide
BroadSoft Partner Configuration Guide Grandstream GXW-400X FXS Analog Gateway & HandyTone HT-50X Analog Telephone Adapter April 2007 Document Version 1.1 BroadWorks Guide Copyright Notice Copyright 2007
Application Notes for BT Wholesale/HIPCOM SIP Trunk Service and Avaya IP Office 8.0 Issue 1.0
Avaya Solution & Interoperability Test Lab Application Notes for BT Wholesale/HIPCOM SIP Trunk Service and Avaya IP Office 8.0 Issue 1.0 Abstract These Application Notes describe the procedures for configuring
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
Application Notes for Configuring Broadvox SIP Trunking with Avaya IP Office - Issue 1.0
Avaya Solution & Interoperability Test Lab Application Notes for Configuring Broadvox SIP Trunking with Avaya IP Office - Issue 1.0 Abstract These Application Notes describe the procedures for configuring
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
Updated Since : 2007-02-18
Microsoft Exchange Server 2007 Unified Messaging PBX Configuration Note: Panasonic KX-TDA200 with AudioCodes MP-11x FXO using Analog lines (In-band DTMF) By : AudioCodes Updated Since : 2007-02-18 READ
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
FACILITY TELECOMMUNICATIONS MANAGEMENT FOR THE GOVERNMENT EMERGENCY TELECOMMUNICATIONS SERVICE Introduction
FACILITY TELECOMMUNICATIONS MANAGEMENT FOR THE GOVERNMENT EMERGENCY TELECOMMUNICATIONS SERVICE Introduction This document provides telecommunications management information for organizations that use the
Voice Mail. Objectives. When you finish this module, you will be able to:
Voice Mail 23 Objectives When you finish this module, you will be able to: Verify that the Embedded Voice Mail (EVM) application can record and play messages. Check the EVM health. Maintain the EVM system.
How To Configure Aastra Clearspan For Aastro (Turbos) And Bpb (Broadworks) On A Pc Or Macbook (Windows) On An Ipa (Windows Xp) On Pc Or Ipa/
BroadSoft Partner Configuration Guide Aastra Clearspan TM April 2011 Document Version 1.4 2811 Internet Blvd. Frisco, TX 75034, U.S.A Tel: 469-365-3237 Fax: 469-365-3071 WWW.AASTRA.COM BroadWorks Guide
Updated Since : 2007-02-09
Microsoft Exchange Server 2007 Unified Messaging PBX Configuration Note: Siemens HiPath3550 with AudioCodes MP-11x FXO using Analog lines (In-band DTMF) By : AudioCodes Updated Since : 2007-02-09 READ
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
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.
Application Notes for G-Tek SIP Telephone MT-102H version 1510X.27.1.02i with Avaya Software Communication System Release 3.0 Issue 1.0.
Avaya Solution & Interoperability Test Lab Application Notes for G-Tek SIP Telephone MT-102H version 1510X.27.1.02i with Avaya Software Communication System Release 3.0 Issue 1.0 Abstract These Application
Mitel MiCloud Telepo for service providers 4.2
Mitel MiCloud Telepo for service providers 4.2 Software for business communication as a service. Key Features For the enterprise market Enable service providers to offer advanced business communication
Xorcom IP-PBX Software Features
Xorcom IP-PBX Software s Based on the Elastix Asterisk i distribution, Xorcom s entire family of IP-PBX appliances provide all the standard telephone functionality supported by Asterisk at no extra cost,
Business Communication Manager BCM 50 and BCM450 Release 5.0 Configuration Guide for Verizon Business SIP Trunking. Issue 1.1
Business Communication Manager BCM 50 and BCM450 Release 5.0 Configuration Guide for Verizon Business SIP Trunking Issue 1.1 Abstract This document provides guidelines for configuring a SIP Trunk between
Application Notes for Configuring Intelepeer SIP Trunking with Avaya IP Office 7.0 - Issue 1.0
Avaya Solution & Interoperability Test Lab Application Notes for Configuring Intelepeer SIP Trunking with Avaya IP Office 7.0 - Issue 1.0 Abstract These Application Notes describe the procedures for configuring
Configuring SIP Trunking and Networking for the NetVanta 7000 Series
61200796L1-29.4E July 2011 Configuration Guide Configuring for the NetVanta 7000 Series This configuration guide describes the configuration and implementation of Session Initiation Protocol (SIP) trunking
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
Implementing Cisco IOS Telephony and Unified Communications Express (IITUCX)
Implementing Cisco IOS Telephony and Unified Communications Express (IITUCX) Course Objectives Explain the benefits and components of a Cisco Unified Communications system Describe how traditional telephony
RT Series IP PBX Products Introduction. All in one telephone system
RT Series IP PBX Products Introduction All in one telephone system RT4 Series RT10 Series RT80 Series RT200 Series Real Tone Technologies Co.,Ltd 7014AB, Tianxia IC Industrial Park,Yiyuan Road Shenzhen
IP PHONE EXPANSION KEY
LICENSABLE Features IP PHONE EXPANSION KEY This phone expansion key enables 8, 16, 32, 64 or 128 additional local IP phone on the QX line of IP PBXs. This key works with a variety of SIP-based endpoints
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
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
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
Updated Since : 2007-02-18
Microsoft Exchange Server 2007 Unified Messaging PBX Configuration Note: Panasonic KX-TES824 with AudioCodes MP-11x FXO using Analog lines (In-band DTMF) By : AudioCodes Updated Since : 2007-02-18 READ
Riverdale City Request for Proposal VoIP Phone System and Unified Messaging
Riverdale City Request for Proposal VoIP Phone System and Unified Messaging Objective Riverdale City (hereinafter referred to as the City ) is currently evaluating VoIP and Unified Messaging platforms
Application Notes for DuVoice with Avaya IP Office 8.1 Issue 1.1
Avaya Solution & Interoperability Test Lab Application Notes for DuVoice with Avaya IP Office 8.1 Issue 1.1 Abstract These Application Notes describe the configuration steps required for the DuVoice hospitality
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
Overview ENUM ENUM. VoIP Introduction (2/2) VoIP Introduction (1/2)
Overview Voice-over over-ip (VoIP) ENUM VoIP Introduction Basic PSTN Concepts and SS7 Old Private Telephony Solutions Internet Telephony and Services VoIP-PSTN Interoperability IP PBX Network Convergence
1 VoIP/PBX Axxess Server
- 1 1 VoIP/PBX Axxess Server The Axxess Server supports comprehensive Voice Over Internet Protocol network services, which are based on the Open Source Asterisk VoIP software. The Axxess Server VoIP telephony
Telco Depot IP-PBX Software Features
Telco Depot IP-PBX Software Features Based on the Elastix Asterisk distribution, Telco Depot s entire family of IP-PBX appliances provide all the standard telephone functionality supported by Asterisk
Silent Monitoring and Recording Using Unified Communications Manager
Silent Monitoring and Recording Using Unified Communications Manager Call centers are expected to guarantee the quality of customer service their agents provide to callers. To that end, the ability to
ADTRAN SBC and Avaya IP Office PBX SIP Trunk Interoperability
6AOSSG001-42B March 2014 Interoperability Guide ADTRAN SBC and Avaya IP Office PBX SIP Trunk Interoperability This guide describes an example configuration used in testing the interoperability of an ADTRAN
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
Mitel MiCloud Telepo for Service Providers 4.0 SP3
Mitel MiCloud Telepo for Service Providers 4.0 SP3 Software for Business Communication as a Service. Key Features For the enterprise market Enable service providers to offer advanced business communication
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
SIP Trunk Configuration Guide. using
SIP Trunk Configuration Guide using www.cbeyond.net 1-877-441-9783 The information contained in this document is specific to setting up SIP connections between Vertical SBX IP 320 and Cbeyond. If you require
Cisco Small Business Unified Communications 300 Series
Cisco Small Business Unified Communications 300 Series Feature Reference Guide January 2011 Introduction The Cisco Small Business Unified Communications 300 Series is a cost-effective, fully featured unified
MITEL SX3300 AX Controller w/ NSU or Internal T1 Card
Avaya Modular Messaging Configuration Note 88035 Version C (7/09) MITEL SX3300 AX Controller w/ NSU or Internal T1 Card T1/QSIG MITEL 3300 AX w/outboard NSU or internal T1 card Avaya MM The PBX and MM
Configuring Interoperability between Avaya IP Office and Avaya Communication Manager
Configuring Interoperability between Avaya IP Office and Avaya Communication Issue 01.01 Contents 1.0 Introduction... 3 1.1 Supported Features... 3 1.2 Network Diagram... 6 1.3 Supported Phones... 6 1.4
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
Avaya Solution & Interoperability Test Lab
Avaya Solution & Interoperability Test Lab Application Notes for Configuring Ascom Wireless IP-DECT SIP Solution with Avaya IP Office 9.0 in a Converged Voice over IP and Data Network - Issue 1.0 Abstract
SIP Trunking using Optimum Business SIP Trunk Adaptor and the Panasonic KX-NCP500 IP PBX V2.0502
PANASONIC SIP Trunking using Optimum Business SIP Trunk Adaptor and the Panasonic KX-NCP500 IP PBX V2.0502 Goal The purpose of this configuration guide is to describe the steps needed to configure the
Software Communication System 500 Release 1.0 System Features
Software Communication System 500 Release 1.0 System Features The Nortel Software Communication System 500 (SCSC500) leads the market in terms of PBX features implemented strictly following the SIP standard.
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
Designed For Market Requirements
Enterprise SIP Designed For Market Requirements Enterprises can combine XO Enterprise SIP with ANY MPLS IP-VPN or Data Network (even from another carrier) for an all-in-one, multi-site IP communications
Http://www.passcert.com
Http://www.passcert.com Exam : 70-337 Title : Enterprise Voice & Online Services with Microsoft Lync Server 2013 Version : DEMO 1 / 18 Topic 1, Litware, Inc Case A Overview Litware, Inc., is an international
Interactive Intelligence CIC 2015 R4 Patch1 Configuration Guide
Interactive Intelligence CIC 2015 R4 Patch1 Configuration Guide Performed By tekvizion PVS, Inc. Contact: 214-242-5900 www.tekvizion.com Copyright 2015 by tekvizion PVS, Inc. All Rights Reserved. Confidential
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
LVS 9000 Overview. 2006 Cisco Systems, Inc. All Rights Reserved. Linksys/Cisco Proprietary All Content Subject To Change Not a Warranty
LVS 9000 Overview 1 Voice over IP is Everywhere Lower Monthly Bills for Broadband Access and the Usage of Broadband Phone Services Coupled with Availability of Advanced Telephony Features Are Driving Small
SIP and H.323. SIP call flow example
SIP and H.323 H.323 Defined by ITU in 1996 International Telecommunication Union Uses many technologies from PSTN A suite of different protocols incl. voice/video SIP Session Initiation Protocol Defined
SIP Trunking using the Optimum Business SIP Trunk adaptor and the AltiGen Max1000 IP PBX version 6.7
SIP Trunking using the Optimum Business SIP Trunk adaptor and the AltiGen Max1000 IP PBX version 6.7 Goal The purpose of this configuration guide is to describe the steps needed to configure the AltiGen
04/09/2007 EP520 IP PBX. 1.1 Overview
1.1 Overview The EP520 IP PBX is an embedded Voice over IP (VoIP) Server with Session Initiation Protocol (SIP) to provide IP extension phone connection for global virtual office of small-to-medium business
DID Destinations are as follows:
IPedge Feature Desc. 1/31/13 OVERVIEW Direct Inward Dialing (DID) Direct Inward Dialing is a feature offered by telephone service providers for use with their customers' PBX systems. The telephone service
Implementing Cisco IOS Telephony and Unified Communications Express (IITUCX)
Implementing Cisco IOS Telephony and Unified Communications Express (IITUCX) Who should attend The primary audience for this course is as follows: Network administrators Network engineers Systems engineers
How To Connect A Phone To An Ip Trunk On A Cell Phone On A Sim Sim Simlia (Vizon) Or Ip Office (Izon) On A Ppl (Telnet) On An Ip Office Softphone On A Vnet (V
Avaya Solution & Interoperability Test Lab Application Notes for SIP Trunking Using Verizon Business IP Trunk SIP Trunk Service and Avaya IP Office Release 7.0 Issue 1.1 Abstract These Application Notes
VIRTUAL COMMUNICATIONS EXPRESS FEATURE DESCRIPTIONS
VIRTUAL COMMUNICATIONS EXPRESS FEATURE DESCRIPTIONS End-User Features: All end users are provided with the features described in this section. Alternate Numbers Enables users to have up to ten phone numbers
Six Questions to Answer When Buying a Phone System
2016 NEW PHONE SYSTEM BUYER S GUIDE Six Questions to Answer When Buying a Phone System In addition to the most comprehensive portfolio of voice communications, data and wireless networking products in
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
