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H T T P : / / W W W . S P H E R E C O M . C O M P A R T N U M B E R V E R S I O N 5 4 0 - 4 0 4 R 7 6 . 2
NOTICES
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Copyright 2008 NEC Sphere Communications Inc., a wholly owned subsidiary of NEC Corporation, Japan. All Rights Reserved. Printed in USA. NEC Sphere Communications Inc. (or Sphere) is continually upgrading and developing the products described in this publication. The information contained herein is specifically designed for the Sphericall v6.2 release and is subject to change without notice. Written permission is required prior to reproduction of any of the work covered here by copyright. For warranty information, see the License Agreement on the Sphericall software DVD media. Sphere, Sphericall, Sphericall Voice Mail, Sphericall Manager, Sphericall Desktop, PhoneHub, COHub, BranchHub, MeetingHub, VG3, and the Sphere logo are trademarks of NEC Sphere Communications Inc. a wholly owned subsidiary of NEC Corporation. NEC is a trademark of NEC Corporation, Japan. Windows, Microsoft, Outlook, and Exchange are trademarks or registered trademarks of Microsoft Corporation. Other products mentioned in this document are the property of their respective owners and are also subject to copyright, trademark, and intellectual property protection as applicable by law. U.S. Patent Numbers 5,892,764 and 6,735,208 and related Foreign Patents. Other U.S. and Foreign Patents Pending.
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540-404r7
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CORPORATE CONTACTS
NEC HEADQUARTERS & ASIA OPERATIONS
NEC Corporation 7-1, Shiba 5-chome Minato-ku, Tokyo 108-8001 Japan Telephone: +81-3-3454-1111 Fax: +81-3-3798-1510 Website: http://www.nec.co.jp/
NORTH AMERICA
NEC Unified Solutions, Inc. 6535 N. State Highway 161 Irving, TX 75039-2402 Telephone: 1-800-240-0632 (800-2400-NEC) Fax: 1-888-318-7932
540-404r7
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Authorized Associates and Resellers: Phone: 1-800-752-6275 Fax: 214-262-5566 Website: http://www.necunified.com
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C ONTENTS
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Contents
Notices ................................................................................................................ iv Document Revision History ............................................................................. iv Contact NEC Sphere Communications................................................................ v Corporate Contacts ......................................................................................... -v
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MGCP IP Phones........................................................................2-5
I. Polycom SoundPoint IP Phone ......................................................................2-5 Planning ............................................................................................................2-6 Final Planning ............................................................................................2-6 Preparing...........................................................................................................2-6 FTP Server, DHCP Server, SNTP Services ..............................................2-6 FTP serverto create login and password for IP phones on FTP server .2-8 FTPto change the IP Phone User Account ............................................2-8 Installing ..........................................................................................................2-11 System Properties .......................................................................................2-11 Sphericall Managerto configure System properties..............................2-11 MGC-To-MGCP Phone Connection Control................................................2-11 Sphericall Managerto configure MGC-to-MGCP phone connections on a Sphere system .........................................................................................2-12 SoundPoint MGCP Phoneto assemble and power the phone .............2-12 To install the MGCP phone with dynamic IP addressing you will need to: ..212 To install the SoundPoint MGCP phone with a static IP address you will need to: .............................................................................................................2-13 Sphericall Managerto complete configuration ......................................2-14 MGCP Phone Installation Test........................................................................2-14 To test for successful extension and station configuration ......................2-14 IP Phone Troubleshooting...............................................................................2-14 Normal Boot Screens ..................................................................................2-15 Installation Issues ........................................................................................2-15 Restarting the Ip Phone...............................................................................2-16 SoundPoint phoneto restart the IP phone ............................................2-16 Sphericall Managerto restart the IP phone remotely ............................2-16 Sphericall Managerto sync IP Phone Files...........................................2-16 Sphericall Managerto Revert to Default Configuration .........................2-16 Sphericall Managerto view Default Parameters ...................................2-17 IP Phone Configuration File Upgrades ........................................................2-17 Convert Polycom MGCP to SIP Command .................................................2-18
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IP Phone Failover ........................................................................................2-19 IP Phone Upgrades.........................................................................................2-19 Using ...............................................................................................................2-19 II. Aastra 480i IP Phone ..................................................................................2-20 Planning ..........................................................................................................2-20 Final Planning ..........................................................................................2-21 Preparing.........................................................................................................2-21 Verify permissions required .....................................................................2-21 FTP serverto create login and password for IP phones on the FTP Server 2-23 FTPto change the IP Phone User Account ..........................................2-23 FTP Serverto add required Sayson files to FTP server........................2-23 TFTP Serverto copy files to the TFTP server.......................................2-23 Sphericall Managerto add FTP server location for use by Aastra phones .. 2-24 Sphericall Managerto add Aastra FTP login and password .................2-24 MGC-To-MGCP Phone Connection Control................................................2-24 Sphericall Managerto configure MGC-to-MGCP phone connections on a Sphere system .........................................................................................2-24 Installing ..........................................................................................................2-25 480i IP Phoneto assemble and power the phone.................................2-25 480i IP Phoneto configure the 480i phone ...........................................2-25 To install the Aastra IP phone with a static IP address you will need to: .2-26 On the phone: ..........................................................................................2-26 Open Internet Explorer:............................................................................2-26 Optional Configurations ...............................................................................2-27 Web Browserto continue using the web browser for optional configurations 2-27 Busy-Lamp-Field Configuration ...................................................................2-28 STEP ONE: Setup Address Group(s) on the Sphericall Manager ...........2-28 STEP TWO: Setup the CTIP Multicast IP Address on the Web Browser 2-28 STEP THREE: Web Browserto configure Busy Lamp Field Keys........2-29 Sphericall Managerto complete configuration ......................................2-30 IP Phone Installation Test ...............................................................................2-30 To test for successful extension and station configuration ......................2-30 Troubleshooting ..............................................................................................2-30 For overview of phone key options: .........................................................2-30 Web Browserto Reset the 480i phone..................................................2-33 Sphericall Managerto Reset the 480i phone from the Sphericall Administrator............................................................................................2-33 480i phonesetto Reset the 480i phone back to factory defaults...........2-33 Changing the Discovery IP Address ...............................................................2-38 STEP ONE: Sphericall Managerto change the Discovery IP Address .2-38 STEP TWO: From the Phoneto change the Discovery IP Address......2-38 IP Phone Upgrades.........................................................................................2-39 Using ...............................................................................................................2-39
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To configure SIP User Agents .................................................................3-45 User Agent Maintenance.................................................................................3-48 To remove a User Agent Parameter ........................................................3-49 SIP Failover.....................................................................................................3-49 create a DNS Record ..................................................................................3-49 On DNS ServerTo create A DNS Record.............................................3-50 Section III - Aastra SIP Phone Configuration ..................................................3-52 Overview .........................................................................................................3-52 To integrate the Aastra phone with the Sphere system ...........................3-52 To enable MWI on an Aastra SIP phone .................................................3-53 Softkeys/Programmable Keys .....................................................................3-54 Hardkeys .....................................................................................................3-54 To save programmable keys from the phone ..........................................3-55 To program keys from the Aastra web interface ......................................3-55 Configuration Files.......................................................................................3-56 Upgrades .....................................................................................................3-56 To upgrade the Aastra Phone..................................................................3-57 Sphere System Upgrades ...........................................................................3-57 Section IV - Grandstream GXP-2000 SIP Phone Configuration..................3-58 Overview .........................................................................................................3-58 Planning ..........................................................................................................3-58 Installing ..........................................................................................................3-59 To integrate the Grandstream GXP-2000 with the Sphere system..........3-59 Web Configuration ..........................................................................................3-59 Account Page ..............................................................................................3-60 Basic Settings Page ....................................................................................3-62 To configure the GXP-2000 basic settings ..............................................3-62 To configure speed dial............................................................................3-63 To configure the time zone ......................................................................3-63 Advanced Settings.......................................................................................3-63 To upgrade the Grandstream GXP-2000.................................................3-64 Grandstream GXV-3000 Phone ......................................................................3-66 Overview .........................................................................................................3-66 Planning ..........................................................................................................3-66 Installing ..........................................................................................................3-67 To integrate the Grandstream GXV-3000 with the Sphere system..........3-67 Web Configuration ..........................................................................................3-67 Account Page ..............................................................................................3-68 Basic Settings Page ....................................................................................3-70 To configure the GXP-3000 basic settings ..............................................3-70 To configure speed dial............................................................................3-71 To configure the time zone ......................................................................3-71 Advanced Settings.......................................................................................3-71 To upgrade the Grandstream GXV-3000.................................................3-72 Section V - Polycom SIP Phone Configuration ...............................................3-74 Planning ..........................................................................................................3-74 Overview .........................................................................................................3-74 Structure of Polycom Configuration Files ....................................................3-75 System Overrides ........................................................................................3-78 Localization..................................................................................................3-79 Convert Polycom MGCP to SIP Command .................................................3-81 Adding a Polycom SIP Phone to A Station..................................................3-81
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CONTEN TS
To add a Polycom SIP phone to the Sphere system ...............................3-81 To add directory information to the Polycom phone ................................3-84 Export Phone Distribution Map....................................................................3-86 To export phone distribution map ............................................................3-86 Upgrades .....................................................................................................3-87 Sphere System Upgrades ...........................................................................3-88 Section VI - UTStarcom F1000G/F3000 SIP Phone Configuration.................3-89 Overview .........................................................................................................3-89 Things To Consider .....................................................................................3-89 A. WiFi and Network Settings......................................................................3-90 On the F1000/3000To configure WiFi and Network Settings ...............3-90 B. F1000/3000 Web Configuration ..............................................................3-91 Web BrowserTo configure the F1000/3000 from the web....................3-91 Web BrowserTo configure F1000/3000 user settings ..........................3-93 Web BrowserTo configure wireless access point settings ...................3-93 C. F1000/3000 and Sphericall Voice Mail ...................................................3-94 Sphericall ManagerTo configure Sphericall Voice Mail settings...........3-94 D. Firmware UpDates ..................................................................................3-94 To update the F1000/3000 firmware version ...........................................3-94 Sphere System Upgrades ...........................................................................3-95 Section VII - SIP Phone Compatibility and Capability with Spherciall Desktop...396 Section VIII - SIP Phone Administrative Star Codes.......................................3-97 Troubleshooting SIP Connections...................................................................3-98
USB Devices...........................................................................5-107
Eutectics IPP200...........................................................................................5-107 Planning.....................................................................................................5-107 Preparing ...................................................................................................5-107 Installing ....................................................................................................5-108 Using .........................................................................................................5-108 Eutectics IPP520...........................................................................................5-108 Planning.....................................................................................................5-108 Overview of Operation...............................................................................5-109 Installation .................................................................................................5-109 Test The Installation ..................................................................................5-109 To test the installation and make sure the IPP520 is properly recognized by
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your PC ..................................................................................................5-109 Configuration .............................................................................................5-109 Functionality ..............................................................................................5-110 Plantronics Wireless set CS50-USB .............................................................5-112 Planning.....................................................................................................5-112 Overview of Operation...............................................................................5-112 Installation .................................................................................................5-112
Music On Hold........................................................................7-123
Music-on-Hold ...............................................................................................7-123 Section I - Hardware-based Music-on-Hold...............................................7-123 To Configure Music-on-Hold Sources....................................................7-125 To assign an extension to a Music-on-Hold station ...............................7-128 Section II - Music-on-Hold Installation Test ...............................................7-129 To test functionality of the Music-on-Hold feature..................................7-129 Section III - Music-on-Hold and Zones ......................................................7-129
Paging .....................................................................................8-133
Paging Lines .................................................................................................8-133 Overview....................................................................................................8-133 Configuring a Paging Line .........................................................................8-134 To add a paging line to a Sphere system ..............................................8-134 To configure a paging line for a Sphere system ....................................8-134 Installing and Integrating a Paging Device ................................................8-136 To integrate a Sphere system with a paging device ..............................8-136 Installation Test .........................................................................................8-138 To verify successful paging system installation and configuration.........8-138
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SMDI Overview .............................................................................................9-139 Manuals .....................................................................................................9-140 SMDI Integration Requirements....................................................................9-140 SMDI Platform Hardware Considerations..................................................9-142 SMDI Operation ............................................................................................9-143 Overview....................................................................................................9-143 Call Processing..........................................................................................9-144 SMDI and System Planning ..........................................................................9-146 Planning for Voice Ports ............................................................................9-146 Call Traffic Calculation: ..........................................................................9-147 Integration Notes .......................................................................................9-149
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To verify the sending of information from the Sphericall Manager.......11-179 To verify the sending of information from the voice mail server...........11-179 Restarts and refreshes ............................................................................11-180 Voice Mail Troubleshooting Steps ...........................................................11-180 Common Voice Mail Issues .....................................................................11-181 Summary.....................................................................................................11-183
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AudioCodes..........................................................................14-203
AudioCodes MP11X Access & Setup .........................................................14-203 Overview of Operation.............................................................................14-203 Choose this method if the installer doesnt have access to the DHCP lease list.........................................................................................................14-203 Choose this method if the installer has access to the DHCP lease list......14204 To update and configure: independent of the method of access .........14-204 Using the AudioCodes FXO MP-11X Quick Setup Screen..................14-205 Using the Automatic Dialing Table.......................................................14-206 Advanced Configuration..............................................................................14-207 CHANNELSELECTMODE* .....................................................................14-207
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ISPROXYUSED.......................................................................................14-207 ProxyName..............................................................................................14-207 PROXYIP.................................................................................................14-208 SIPGATEWAYNAME ..............................................................................14-208 ISREGISTERNEEDED............................................................................14-208 REGISTRARNAME .................................................................................14-208 REGISTRATIONTIME .............................................................................14-208 uSERname ..............................................................................................14-209 AuthenticationMODE ...............................................................................14-209 CODERNAME .........................................................................................14-209 ENABLECURRENTDISCONNECT .........................................................14-209 Disconnectonbrokenconnection ..............................................................14-209 Currentdisconnectduration ......................................................................14-210 TimeToSampleAnalogLineVoltage ..........................................................14-210 enablecallerID..........................................................................................14-210 pSTNPREFIX ..........................................................................................14-210 TRUNKGROUP_1 ...................................................................................14-210 enablecallerid_<Port>..............................................................................14-210 targetofchannel<Port> .............................................................................14-211 ISTWOSTSTAGEDIAL ............................................................................14-211 ISWAITFORDIALTONe ...........................................................................14-211 DTMFTransportType ...............................................................................14-211 MFTransportType ....................................................................................14-211 AudioCodes MP104 Access & Setup..........................................................14-212 Planning ......................................................................................................14-212 Overview of Operation.............................................................................14-212 Choose this method if the installer doesnt have access to the DHCP lease list.........................................................................................................14-212 Choose this method if the installer has access to the DHCP lease list......14213 To update and configure: independent of the method of access .........14-213 Using the AudioCodes FXO MP104 Quick Setup Screen ...................14-214 Fax Signaling Method Using T.38........................................................14-215 Using the Automatic Dialing Table.......................................................14-215 Advanced Configuration..............................................................................14-216 CHANNELSELECTMODE* .....................................................................14-216 ISPROXYUSED.......................................................................................14-216 ProxyName..............................................................................................14-216 PROXYIP.................................................................................................14-217 SIPGATEWAYNAME ..............................................................................14-217 ISREGISTERNEEDED............................................................................14-217 REGISTRARNAME .................................................................................14-217 REGISTRATIONTIME .............................................................................14-218 uSERname ..............................................................................................14-218 AuthenticationMODE ...............................................................................14-218 CODERNAME .........................................................................................14-218 ENABLECURRENTDISCONNECT .........................................................14-218 Currentdisconnectduration ......................................................................14-218 TimeToSampleAnalogLineVoltage ..........................................................14-219 enablecallerID..........................................................................................14-219 pSTNPREFIX ..........................................................................................14-219 TRUNKGROUP_1 ...................................................................................14-219
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enablecallerid_<Port>..............................................................................14-219 targetofchannel<Port> .............................................................................14-220 ISTWOSTSTAGEDIAL ............................................................................14-220 ISWAITFORDIALTONe ...........................................................................14-220 DTMFTransportType ...............................................................................14-220 MFTransportType ....................................................................................14-220
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SIP Trunking.........................................................................17-231
Overview of SIP ..........................................................................................17-231 SIP Terminal Location in Sphericall ............................................................17-232 NEC Sphere SIP Trunking ..........................................................................17-233 Before You Begin Any SIP Integration ....................................................17-234 To configure SIP User Agents .............................................................17-234 User Agent Maintenance.............................................................................17-237 To remove a User Agent Parameter ....................................................17-238 SIP Trunking to SIP Service Provider .........................................................17-239 To plan to install SIP trunking ..............................................................17-240 Before you begin the trunk configuration .............................................17-241 To configure a SIP trunk ......................................................................17-241 To configure Service Provider information for the softtrunk.................17-245 To configure SIP trunk properties ........................................................17-246 Vendor Specific SIP Trunking Configuration........................................17-252 SIP Trunking Tie Line..................................................................................17-253 To install SIP trunking or tie lines.........................................................17-253 To configure SIP User Agents .............................................................17-254 To configure SIP for tie line..................................................................17-255 To configure SIP trunk properties ........................................................17-260 To verify the softtrunk is registered on the far end of tie line ...............17-263 SIP tie line to Third-Party App.....................................................................17-266 To install SIP trunking or tie lines.........................................................17-266 To configure SIP ..................................................................................17-267 To configure SIP trunk general properties ...........................................17-271 Troubleshooting SIP Connections...............................................................17-276
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Configuring Survivable Outbound Routing (Survivable models) .................18-294 To enable survivable outbound routing................................................18-294 To load the file onto the Quintum gateway ..........................................18-294 Understanding the Environment..................................................................18-295 Testing ........................................................................................................18-296
Index.............................................................................................I-1
Document Feedback ...........................................................................................-1 Document Information .....................................................................................-1
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The Sphericall telecommunications software suite delivers integrated communications to the enterprise desktop. Sphericall provides a foundation of traditional PBX capabilities combined with desktop video, text messaging, on-line phone book, status presence controls, on-demand conference bridging, IP-phone synchronization, and integration with enterprise tools such as Exchange & Outlook. MGCP, SIP Standards-based VOIP Protocols SIMPLE messaging protocol - Windows Messenger Supports range of industry leading IP Telephones Multiple enterprise and SOHO gateway options General SIP Configuration Notes
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IN THIS MANUAL
In this manual, you will find information on how to integrate several third-party products into the Sphere system. Our suite of Alliance products prove that the Sphere system is more than adequate for meeting your telecommunications needs over the network. All end points on the network can take advantage of the Sphericall Desktop for multiple forms of communication including voice, video, and text messaging, in addition to Exchange and Outlook. Alliance products also include SNMP and SMDI integrations, Music-on-Hold and Paging integrations, and the Windows Messenger phone client.
DOCUMENT INDEX
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For complete instructions on installing the Sphere system, refer to our documentation. The documentation is listed here in the order in which a system is installed.
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I N T E G R A T E P A R T N E R TE C H N O L O G I E S
Document Index
Sphere Document Name: Release Notes & Upgrade Procedures Sphere System Requirements Sphere Star Codes
Document Description
Release Notes for current product & upgrade procedures System Requirements, minimums, limits, requisites Telephone Set, Administrative and Diagnostic Star Codes
Book 1
Project Planning & System Requirements; Site Walk Workbook; Customer Implementation; Import & Export Utility; Stage & Cut Information; Logins & Passwords Required for Installation; Planning for SIP Installations; Network Planning Information; Suppliers Full information and process for a new installation everything from software installation to detailed integration including: Install Sphericall Software; System Properties; LANs; Number Plan; Telephony Area Settings; Station Template Settings; Station Port Settings; PSTN Settings; Mapping Lists; Secondary Managers; User Rights & Permissions for Sphericall Desktop; Sphericall Desktop & Softphone; ARS; Media Server Options; Optional Connections; Call Recording; Call Admission Control; Optional Desktop Connections Planning, installing and troubleshooting Sphericall Voice Mail (Exchange 2000/2003); Auto Attendant Configuration Troubleshooting; Theory of Operation; Sphericall Voice Mail User Quick Reference Content may vary depending on third-party partners. All third-party product integrations are documented for reference Moves, Adds & Changes to the System; Daily Management; Sphere Services & Utilities; Supporting Sphericall Desktop; Reports, Statistics and Tools; Support Resources; User, System & Hardware Changes; Emergency Backup Plans; Additional Support Resources, Quick Reference Charts, Settings, Microsoft Resources Emergency Service Planning, Installation and setup; MLPP or CallNOW implementation Using the telephone Sphericall Desktop Software Sphericall Desktop Options Guides to aid in use of Sphericall
Book 2
Book 3
Book 4
Book 5
Sphericall Desktop Quick Reference Guide Sphericall Voice Mail Quick Reference Guide English Sphericall Voice Mail Quick Reference Guide Spanish (Mexico)
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I N T E G R A T E P A R T N E R TE C H N O L O G I E S
Document Index
Document Description
Reference Guides related to Sphericall integration with partner products
COHub Manual PhoneHub Manual BranchHub Manual MeetingHub Manual MG CLI Reference Manual
Admin Help Phone Help Visual Basic ComAPI Help NEC Sphere document(s) with all the features listed
Install & Configure; Troubleshooting; Display Messages; Specifications Install & Configure; Troubleshooting; Display Messages; Specifications; Emergency Failover Install & Configure; Troubleshooting; Display Messages; Specifications; Emergency Failover Install & Configure; Troubleshooting; Display Messages; Specifications A complete reference to the Media Gateway Command Line Interface; SNMP traps Searchable help file with Admin Help only Searchable help file of Desktop Help only Help file for generating VB com objects TBD 2008
located in CD:\\...Documents\Document_INDEX.pdf Must run the index on a local PC to all document folders, then place it on the Installs\Documents\version folder Source files are required in that folder
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I N T E G R A T E P A R T N E R TE C H N O L O G I E S
Document Index
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MGCP IP PHONES
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This chapter supports the following MGCP IP phones: I. Polycom SoundPoint IP phone II. Aastra 480i IP phone
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The SoundPoint IP phone is a standards-based internet protocol phone that delivers rich applications to the desktop as part of the Sphere system. The phone is from Polycom, a leader in advanced telephony. Platform-independent design enables a seamless interface to all leading protocols and platforms, making the SoundPoint the perfect complement to the Sphere system. The SoundPoint is a full-featured, intelligent endpoint on the converged network. IP phones connect to your Ethernet network. A single Cat 5 connection to the desk serves both the PC and phone. The SoundPoint provides excellent voice quality with the latest microprocessor technology and support for quality-of-service processing. Programmable feature keys and context-sensitive keys let you access Sphericall advanced functions with a simple touch. It also features speakerphone functionality and an extra large graphic display for time, date, caller ID data and future data streams. Failover The SoundPoint IP phone supports failover. If a Primary Sphericall Manager becomes unavailable, all IP phones automatically fail over to the Secondary Sphericall Manager (if one exists within that Sphere system). When the Primary Sphericall Manager becomes available, the phones will automatically move back.
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MGCP IP PHONES
Planning
PLANNING
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The following information needs to be considered before installing an IP phone: Each IP phone needs three addresses in order to begin its configuration: IP address FTP server address SNTP server address
Note: Microsoft Windows Server has built-in DHCP and FTP servers which ease the
installation process. Sphere recommends that the FTP server is not housed on the Primary Sphericall Manager. Make sure an FTP server is set up on your network. For notes related to configuration of FTP, SNTP or DHCP see Appendix A of this manual, or refer to Microsoft documentation.
Note: Every system requires adherence to the Sphere System Requirements.
Final Planning
1 2
Review Sphere System Requirements for domain password issues. Verify that the rest of the system is operational prior to starting voice mail services.
PREPARING
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FTP Server, DHCP Server, SNTP Services
1 2 3 4
Install Static or Dynamic IP Address components to the Sphere system. For more information, review Appendix A of this manual. Setup SNTP services for the Sphere system. Verify FTP server on the network. Verify permissions required.
Table 2.1
Account Logins/Group Needed: Group Policy
Account Creation:
Support for Microsoft Windows Installer and Group Policy Snap-in for installing, updating or uninstalling Sphericall Desktop or Sphericall Desktop Softphone clients. See Microsoft product documentation for full permission information.
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MGCP IP PHONES
Preparing
Account Creation:
Organizations using a separate FTP server must do the following: Copy the FTProot directory from the Sphericall Manager to the FTP Server. Grant FULL CONTROL Security access to the FTProot directory, based on the type of phone(s) you are using (i.e. if using SoundPoint IP phones, you must create that account; if using Aastra 480i phones, you must create that account; if using both, you must create both accounts). Required Folder(s): If you are installing Windows FTP server, the ftproot folder will be located by default at: c:\inetpub\ftproot. This default setting needs to be changed as follows: For systems with the FTP Server on the Primary Sphericall Manager, the following folder is required for the location of IP phone resource files: <drive>:\\Program Files\Sphere\ftproot For systems with the FTP Server on any other server (third-party or Secondary Sphericall Manager), the following folder is required for the location of IP phone resource files: <drive>:\\ftproot\ Aastra 480i FTP login: Login: Sayson; Password: Aastra480i Either of the following two: 1. Must be a Domain User if FTP server is also the Domain/Active Directory server. 2. Local User if on the FTP server and FTP server is not the Domain/Active Directory server. Create a user account with username Sayson; create password: Aastra480i Default Administrative Passcode on 480i phone: 22222 Configurable Administrative Passcode on 480i phone only (not web interface): 1) Options 2) Option #9 - MGCP Settings 3) Option #9 - Admin Password Administrative Passcode & web interface login on 480i phones: Login: admin Password: 22222 Default Administrative Passcode on 9112i and 9133i phone: 22222
Aastra480i
Administrative Passcode & web interface login on 9112i and 9133i phones: Login: admin Password: 22222 Default administrative password: admin
PlcmSpIp
(case sensitive)
See Active Directory User Accounts Notes Below
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MGCP IP PHONES
Preparing
Account Creation:
CREATE A LOCAL USER ACCOUNT (WITH PASSWORD) FOR THE IP PHONES ON THE FTP SERVER
FTP serverto create login and password for IP phones on FTP server
1 2 3 4 5
Create a local user account on the FTP server with username PlcmSpIp.
Create password PlcmSpIp. Deselect User must change password at next logon. Select User cannot change password. Select Password never expires.
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MGCP IP PHONES
Preparing
2-9
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MGCP IP PHONES
Preparing
Figure 2.2
Figure 2.3
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MGCP IP PHONES
Installing
INSTALLING
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SYSTEM PROPERTIES
Before configuring your organizations IP phones, you need to define certain characteristics of the Sphere system. These settings will dictate functionality of the connected MGs and IP phones.
Type the name or address (or Browse to select) of the FTP server used to store the XML configuration files for your organizations IP phones in the Server Name field.
IP phones on a Sphere system must download XML configuration files from an FTP server on the network. These files are installed into the FTP root directory upon Sphere system installation and are responsible for setting the functional parameters of the individual IP phones.
5 6 7 8 9
Click the IP Phones tab. Verify the login name to the FTP server in the Polycom Login Name field. Verify the password for access to the FTP server in the Polycom Password field. Click Apply. Click OK.
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MGCP IP PHONES
Installing
Click the General tab. Expand Media Gateway Controllers from the tree list. Highlight the Media Gateway Controller you wish to configure. Right-click to View Properties. Click the MGCP Phones tab.
From the Properties for Media Gateway Controllers window: In the Set As Default area:
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Select the Primary MGC check box if this Sphericall Manager is to be considered the default Primary Sphericall Manager that THIS MGCP phone will use to obtain its configuration information upon initialization. Note: Settings configured for individual MGCP phones will override this default
setting.
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Select the Secondary MGC check box if this Sphericall Manager is to be considered the default Secondary Sphericall Manager THIS MGCP phone will use to obtain its configuration information upon initialization if the Primary Sphericall Manager is unavailable. Note: Settings configured for individual MGCP phones will override this default
setting.
8 9
Click ADD under each category to add this MGCP phone to its Primary, Secondary or Tertiary MGC. Click Apply.
You will receive a message asking if you want to restart the MGCP phones. Select the appropriate option based on your system. The MGCP phone will need to be restarted to apply all these settings.
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Click OK.
You may also choose the MGC to which each MGCP phone will be assigned via their Station Properites.
Please use the quick start guide provided with your SoundPoint phone for phone assembly and cable connections.
See the above section for the keycap layout of the phones.
To install the MGCP phone with dynamic IP addressing you will need to:
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MGCP IP PHONES
Installing
As the phone initializes, a Welcome screen is displayed with a count-down timer and three softkeys: START, SETUP, and ABOUT.
2 3
Press the SETUP softkey. A menu is displayed on the phone screen. Use the arrow keys on the upper right of the phone to scroll through the menu options.
You MUST press the EDIT softkey to make configuration changes and the SAVE softkey to accept changes to that option.
4
Configure the DHCP server with the appropriate scope and information to pass to the various endpoints on the network:
IP address range Default gateway and subnet mask addresses FTP server address SNTP server address SNTP time offset (GMT offset) When the MGCP phone initializes, it broadcasts to locate a DHCP server. The DHCP server will pass the IP address, FTP server address, and SNTP server address to the phone based upon the configuration of the DHCP scope. The MGCP phone re-initializes and downloads a generic XML file. This file contains the information necessary for normal runtime operations including the location of the MGC(s) to which this phone will look for address identification. After the MGCP phone re-initializes, the MGCP phone appears in the Sphericall Administration application.
To install the SoundPoint MGCP phone with a static IP address you will need to:
1
As the phone initializes, a Welcome screen is displayed with a count-down timer and three softkeys: START, SETUP, and ABOUT.
2 3
Press the SETUP softkey. A menu is displayed on the phone screen. Use the arrow keys on the upper right of the phone to scroll through the menu options. Note: You MUST press the EDIT softkey to make configuration changes and the
IP address of the MGCP phone Default gateway and subnet mask addresses FTP server address SNTP server address SNTP time offset (GMT offset)
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MGCP IP PHONES
The MGCP phone re-initializes and downloads a generic XML file from the FTP server. This file contains the information necessary for normal runtime operations including the location of the MGC(s) to which this phone will look for address identification. After the MGCP phone re-initializes, the MGCP phone appears in the Sphericall Administration application.
Return to the Properties for Station and Properties for Extension windows to configure settings including
Forwarding conditions Emergency group Telephony area association (for stations) and hunt order settings (for extensions). Refer to Book 2: Install & Configure Chapter 6, Station Local Settings.
...........................................................
Make calls between MGCP phones. Make calls between MGCP phones and other types of phones or external calls. Test MGCP phone keys and softkeys.
IP PHONE TROUBLESHOOTING
...........................................................
Most troubleshooting issues may arise during the initial deployment of SoundPoint IP phones versus when the system is operational. It is essential to remember that three key elements are necessary to deploy the SoundPoint IP phones: IP address Either assign a static IP address or have an operational DHCP server with a properly defined DHCP scope FTP server address You MUST have an operational and properly configured FTP server SNTP server address The phone MUST get its date and time from an operational FTP server
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MGCP IP PHONES
IP Phone Troubleshooting
8.
INSTALLATION ISSUES
During the boot sequence of the IP phone, the following error messages may be shown.
Table 2.3
Error Message
Resolution
1. Verify your DHCP server is operational and connected to the network. 2. If you are using static IP addressing, disable DHCP Client on the phone and enter the appropriate IP address and Default Gateway. Scenario One 1. Verify that the <MAC>.cfg or 00000000000.cfg file is in the root directory of the FTP user account. Scenario Two 1. Verify the FTP settings in the DHCP scope are correct. 2. Verify that the FTP server is operational and connected to the network. 3. Verify the FTP user and password match on both the phone and the FTP server.
Failed to get boot parameters via DHCP. Tell your SysAdmin, 500 seconds until autoboot.
There are two scenarios when you might see this error: 1. The FTP server was found and logon successful. The <MAC>.cfg was not found on server. The 00000000000.cfg file was not found on server. The <MAC>.cfg was not found in FLASH. 2. The FTP server was NOT found or login was unsuccessful, and the <MAC>.cfg was not found in flash.
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MGCP IP PHONES
IP Phone Troubleshooting
Error Message
Description
This error message indicates when there has been a loss of connection to the IP switch or the switch has lost its network connection.
Resolution
1. Verify your IP cable connection to the IP switch. 2. Verify the switch connection to the rest of the LAN.
Note: PING is a useful tool when trying to determine connectivity of DHCP, FTP, or
IP501 & IP 600 Press the four following keys simultaneously until the IP phone restarts:
Both Audio Path Volume keys (+ -) Hold key Voice Mail key simultaneously until the IP phone restarts.
IP301 Press the four following keys simultaneously until the IP phone restarts:
Both Audio Path Volume keys (+ -) Hold key Directories key simultaneously until the IP phone restarts.
Expand the Stations tree in this window view. Click to highlight the IP phone unit. Right-click. Select Restart.
Click Tools\Configure MGCP Phones\Polycom\Sync IP Phone Files. Confirm that you want to complete this task by clicking YES.
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MGCP IP PHONES
IP Phone Troubleshooting
The Sphericall Administrator application will backup the current PlcmSpIp files in the PlcmSplp\backup directory. The Sphericall Administrator application will check to see if the SPHERICALL_VERSION tags within the Sphericall.cfg and 000000000000.cfg files are newer than the current file tags. If so, the PlcmSpIp\Update\000000000000.cfg file is copied into PlcmSpIp\000000000000.cfg. The PlcmSpIp\Update\Sphericall.cfg is copied into PlcmSpIp\Sphericall.cfg. It is then updated (information gathered from the database) with the Sphere systems current default Primary and Secondary Sphericall Manager(s). If 000000000000.cfg was changed, the MAC-specific.cfg (for example, 0004f20011110.cfg) files are updated. All other files remaining in the PlcmSpIp\Update directory are copied into the PlcmSpIp directory unchanged. The update directory contents are deleted.
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MGCP IP PHONES
IP Phone Troubleshooting
Figure 2.5
Admin window
the IP 300 and IP 500 products. These products were discontinued in May 2006 (Refer to Product Bulletin Number 532.PB available from The Polycom Resource Centre (http://extranet.polycom.com). They were replaced with the SoundPoint IP 301 and SoundPoint IP 501 products which have additional internal memory.
Note: Installations that have a mixture of IP 300 and/or IP500 phones deployed
along with other models will require changes to the phone configuration files to continue to support IP 300 and IP 500s when software releases SIP 2.2 and newer are deployed. Please refer to Polycom knowledge base articles for further and updated information.
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MGCP IP PHONES
IP Phone Upgrades
Figure 2.6
IP PHONE FAILOVER
IP phones within a Sphere system can be configured, via the Sphericall Administration application, to fail over to the appropriate Sphericall Manager (from their Primary Sphericall Manager). If a failover occurs, the IP phone tries to register with the Primary Sphericall Manager. When the phone discovers that the Primary Sphericall Manager is not available, the phone fails over to the Secondary Sphericall Manager.
IP PHONE UPGRADES
...........................................................
All upgrades related to IP phones generate from Sphericall software upgrades. If there is a full system upgrade, IP phones will be upgraded as a part of this process. Refer to the Sphere System Release Notes for full upgrade procedures.
USING
...........................................................
The Quick Reference Guide for this version of IP phone is located in the Appendix of this manual.
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MGCP IP PHONES
...........................................................
Sayson and Aastra Technologies Ltd., have developed a versatile, feature-rich IP screenphone that supports current and emerging industry standards. The 480i has been designed to work with the Sphere system. Saysons large, backlit display, with 5 programmable "Softkeys," allow for Sphericall and Sayson technology integration. A 10/100 Ethernet switch eliminates the need for additional wiring to the desktop.
PLANNING
...........................................................
The following information needs to be considered before installing an Aastra 480i phone: Each IP phone needs three addresses in order to begin its configuration: IP address FTP server address SNTP server address
Note: Microsoft Windows Server has built-in DHCP and FTP servers which ease the
installation process. Sphere recommends that the FTP server is not housed on the Primary Sphericall Manager. Make sure an FTP server is set up on your network. For notes related to configuration of FTP, SNTP or DHCP see Appendix A of this manual, or refer to Microsoft documentation.
Note: Every system requires adherence to the Sphere System Requirements.
When an IP phone initializes, DHCP is enabled by default. The DHCP server passes information to the IP phone so that it can configure itself for subsequent Sphericall address assignment and normal runtime operations. If you are planning on using Dynamic IP addresses, make sure a DHCP server is set up on your network.
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MGCP IP PHONES
Preparing
If you are not planning on using a Dynamic IP address, refer to the section Configuring the 480i IP Phone on how to set up an IP Address manually on each phone. Caution: If you choose to use static IP addresses with your 480i implementation, you will not have designed your system for failover capability. Failover requires the use of Dynamic IP Addresses.
Final Planning
1 2
Review Sphere System Requirements for domain password issues. Verify that the rest of the system is operational prior to starting voice mail services.
PREPARING
Verif y permissions required
Table 2.4
Account Logins/Group Needed: Group Policy
...........................................................
Third-party permissions required
Account Creation:
Support for Microsoft Windows Installer and Group Policy Snap-in for installing, updating or uninstalling Sphericall Desktop or Sphericall Desktop Softphone clients. See Microsoft product documentation for full permission information. Organizations using a separate FTP server must do the following: Copy the FTProot directory from the Sphericall Manager to the FTP Server. Grant FULL CONTROL Security access to the FTProot directory, based on the type of phone(s) you are using (i.e. if using SoundPoint IP phones, you must create that account; if using Aastra 480i phones, you must create that account; if using both, you must create both accounts). Required Folder(s): If you are installing Windows FTP server, the ftproot folder will be located by default at: c:\inetpub\ftproot. This default setting needs to be changed as follows: For systems with the FTP Server on the Primary Sphericall Manager, the following folder is required for the location of IP phone resource files: <drive>:\\Program Files\Sphere\ftproot For systems with the FTP Server on any other server (third-party or Secondary Sphericall Manager), the following folder is required for the location of IP phone resource files: <drive>:\\ftproot\
Domain Administrator permissions required for Group Policy Administration See below based on manufacturer
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MGCP IP PHONES
Preparing
Account Creation:
Aastra 480i FTP login: Login: Sayson; Password: Aastra480i Either of the following two: 1. Must be a Domain User if FTP server is also the Domain/Active Directory server. 2. Local User if on the FTP server and FTP server is not the Domain/Active Directory server. Create a user account with username Sayson; create password: Aastra480i Default Administrative Passcode on 480i phone: 22222 Configurable Administrative Passcode on 480i phone only (not web interface): 1) Options 2) Option #9 - MGCP Settings 3) Option #9 - Admin Password Administrative Passcode & web interface login on 480i phones: Login: admin Password: 22222 Default Administrative Passcode on 9112i and 9133i phone: 22222
Aastra480i
Administrative Passcode & web interface login on 9112i and 9133i phones: Login: admin Password: 22222 Default administrative password: admin
PlcmSpIp
(case sensitive)
See Active Directory User Accounts Notes Below
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MGCP IP PHONES
Preparing
CREATE THE LOCAL USER ACCOUNT (WITH PASSWORD) FOR THE AASTRA 480I IP PHONES ON THE FTP SERVER
FTP serverto create login and password for IP phones on the FTP Server
1 2 3 4 5
Create a local user account on the FTP server with username Sayson.
Create password Aastra480i. Deselect User must change password at next logon. Select User cannot change password. Select Password never expires.
Install 480i specific XML file for screen management on FTP server. Note: Installing an FTP server on your Primary Sphericall Manager is not
recommended. If the FTP server resides on one of the Sphericall Managers within your system, you do not have to place these files on the FTP server. The Sphericall Manager will discover them automatically. The appropriate path to the FTP site directory for a Sphere system integrating the Aastra 480i phone is: [Sphericall Manager hard drive]:\program files\sphere\ftproot\sayson\SaysonDeck1.xml This is the directory in which all XML configuration files will be stored within the Sphere system for use by the 480i phones. If this file is newer than the file the phone is currently using, the phone will download this file from the server.
2
If you have a separate FTP server, paste the SaysonDeck1.xml file into the following FTP directory on the FTP server:
Files stored in the Sphericall Manager for the TFTP server are:
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MGCP IP PHONES
Preparing
c:\program files\sphere\images\appmgcp.bin.gz File 1. 480i.image File 2. 480i.update-flash.image Every Sphericall Manager is a TFTP server and is updated with these files during installation or upgrade.
Sphericall Managerto add FTP server location for use by Aastra phones
1 2 3 4
Right-click on the System utility from the General tab of the Sphericall Administrator application. View Properties. Select the General tab. Enter or browse to select the FTP Server name in the FTP Server section.
Right-click on System from the General tab. View Properties. Select IP Phones tab. Verify the following login/password: Login: Sayson Password: Aastra480i
Click the General tab. Expand Media Gateway Controllers from the tree list. Highlight the Media Gateway Controller you wish to configure. Right-click to View Properties. Click the MGCP Phones tab.
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MGCP IP PHONES
Installing
From the Properties for Media Gateway Controllers window: In the Set As Default area:
6
Select the Primary MGC check box if this Sphericall Manager is to be considered the default Primary Sphericall Manager that THIS MGCP phone will use to obtain its configuration information upon initialization. Note: Settings configured for individual MGCP phones will override this default
setting.
7
Select the Secondary MGC check box if this Sphericall Manager is to be considered the default Secondary Sphericall Manager THIS MGCP phone will use to obtain its configuration information upon initialization if the Primary Sphericall Manager is unavailable. Note: Settings configured for individual MGCP phones will override this default
setting.
8 9
Click ADD under each category to add this MGCP phone to its Primary, Secondary or Tertiary MGC. Click Apply.
You will receive a message asking if you want to restart the MGCP phones. Select the appropriate option based on your system. The MGCP phone will need to be restarted to apply all these settings.
10
Click OK.
You may also choose the MGC to which each MGCP phone will be assigned via their Station Properites.
INSTALLING
...........................................................
480i IP Phoneto assemble and power the phone
Please use the Installation Guide provided with your phone for phone assembly and cable connections.
1 2
Complete all cable and cord connections for operability. Refer to the Troubleshooting section of the chapter for an overview of the phone key setup options.
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MGCP IP PHONES
Installing
Options: select 8 Network Select 1 DHCP (press change until Yes appears under DHCP)
2
Determine IP Address Options: select 8 Network Select 2 IP Address (IP address will be displayed) Make note of IP address
Once an IP address has been assigned to a phone, you may now configure with the Options keys on the phoneset, or via the Sayson Web Client from any web browser.
3
The factory defaults on the phone are now in place. Important: Once an IP address has been assigned to the phone, you may see the time is not correct. Restarting the phone at this juncture allows the 480i to check-in with the Sphericall Manager and receive its time stamp from that server.
Aastra 480i factory installation guide within the packaging covers how to reset the offset time from Eastern time to your timezone. Sphere also provides this manual on the Documentation folder on the DVD. Select the Options button on the phone. Choose Option 2 - Time and Date. Choose Option 6 - Time Zones. Choose the appropriate time zone.
4
To install the Aastra IP phone with a static IP address you will need to:
After connecting phone to network:
On the phone:
1 2 3 4 5 6 7 8 9 10 11
Select Options. Scroll to Network (Option 8). You will be required to login. Select Enter. Select Show for DHCP (Option 1). Press change to select No for DHCP. Select Done. Scroll to select Option 2 (IP Address). Press Show and enter IP Address. Press Done. Press Done again until you return to the No Service Screen.
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MGCP IP PHONES
Installing
Complete the following fields based on your IP addresses for your system:
Click Set Values. Click Admin Firmware. Enter the IP address if the FTP Server. Click Call Client. Enter information:
Call Agent Address 1 = Primary MGC Call Agent Address 2 = Any Additional Managers Call Agent Address 3 = Any Additional Managers Call Agent Port: 2727 Client Port: 2427 FTP User Name: Sayson FTP Password: Aastra480i CTIP: Enter information from Sphericall Administrator Configuration. Discovery: 239.193.0.0
10 11
OPTIONAL CONFIGURATIONS
Web Browserto continue using the web browser for optional configurations
1
Enter the IP address of the phone into the web browser address field.
Example: 192.168.0.100
2 3
Press enter. Logon as administrator: Login: admin Password: 22222 View default configuration of 480i phone.
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MGCP IP PHONES
Installing
If you are using the Sphere Discovery Protocol, there will be minimal changes needed to the defaults. Consider the following changes as per your dial plan and system design:
Busy Lamp Field Entries in this field allow users to monitor other phone lines. (BLF) CTIP Multicast A CTIP Multicast IP Address and Port is needed in the Address Group
on the Sphere system in order for users to monitor other users with Sphericall Desktop, Administrator, or with the 480i BLF field. If an administrator wants to enable monitoring by Zone, the CTIP Multicast IP Address from the Zone will need to be entered into this field on the Aastra Web Client or via options on the phone.
Reset User System administrators may reset the user password to allow individual Password users to enter their own BLF fields. This is used on a case-by-case
basis as system administrators deem appropriate.
BUSY-LAMP-FIELD CONFIGURATION
Administrators may want to configure some phones to display the extensions of monitored lines. These optional fields are configured via the web browser interface. Sphere system administrators may enter users BLF extensions for monitoring, or they may allow users to enter their own extensions for monitoring. If they allow users the right to enter their own extensions, they will need to provide the user with 1) the IP address of the 480i phone, 2) the user login and password: login = user, password = nothing (do not enter anything into the field). Sphere recommends a three step process for preparing users to monitor from BusyLamp-Field (BLF):
From the Sphericall Administrator application, right-click on Address Groups from the General tab. Click Add. Fill in the name of the Address Group (Ex: 480i phone monitoring, etc.). Select the appropriate Multicast Address for this purpose, or create a new address. Click Add in the lower section to add users/stations to this Address Group.
Only users who have been added to an Address Group will be able to monitor other users using the 480i BLF feature.
6
STEP TWO: Setup the CTIP Multicast IP Address on the Web Browser
1
2-28
MGCP IP PHONES
Installing
Only those with admin login privileges may change the CTIP Multicast IP Address field.
2 3 4
Select Call Client. Enter the appropriate CTIP Multicast IP Address from the Address Group on Sphericall into this field. If using more than one CTIP Multicast IP Address for different users, logon to the Aastra Web Browser as Administrator and enter a new users IP address in the browser address. Enter a new/different CTIP Multicast IP Address for this user. Repeat for all users or for all unique Address Groups.
Select Soft Keys from the Web Browser User field. Enter extensions and names of users this phone will monitor using the BLF feature.
Figure 2.7
In addition to listing the monitored extensions, the following states may shown on the phone:
Table 2.5 Busy Lamp Field States
State
Idle
Icon
Ringing
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MGCP IP PHONES
State
Connected
Icon
Do Not Disturb
Call Forwarded
Return to the Properties for Station and Properties for Extension windows to configure settings including
Forwarding conditions Emergency group Telephony area association (for stations) and hunt order settings (for extensions). Refer to Book 2: Install & Configure Chapter 6, Station Local Settings.
...........................................................
Make calls between IP phones. Make calls between IP phones and other types of phones or external calls. Test IP phone keys and softkeys.
TROUBLESHOOTING
For overview of phone key options:
...........................................................
Administrators have two options for setting up 480i phones. They may use the Options buttons on the phoneset or they may use the web browser.
2-30
MGCP IP PHONES
Troubleshooting
Options refers to the recessed button on the 480i phone used for configuration.
Press Options to enter the Options list.
Options
Use the up or down arrow buttons to scroll through the list of options
Press Show softkey and arrow buttons to select an option. Press the Done softkey at any time to exit the option and save the change. Press the Cancel softkey at any time to exit without saving changes.
From the phoneset, there are Administrator level options that are secured by a passcode. The default administrative passcode is: 22222. However, you have the ability to change the default administrative passcode from: Option 9: MGCP Settings Option 9: Admin Password The following Options are those which may be configured at install: Please note each of these options may also be configured, verified or changed from the web browser as well.
Option 8
Network Settings 1. DHCP Turns DHCP on or off. IP Address, Subnet Mask and Gateway options are read only when DHCP is on. Default: DHCP is on. 2. IP Address 3. Subnet Mask 4. Gateway 5. DNS 6. TFTP TFTP server address is where the configuration files will be download from.
Option 9
MGCP Settings
If DHCP is turned on and the Sphere Discovery Protocol is enabled, all the MGCP settings will be automatically configured. You should only have to set these if the Discovery Protocol has not been enabled. 1. Call Agents This option contains up to 3 Call Agent IP Addresses. The Discovery Protocol automatically configures this setting. If the Discovery IP option is configured correctly, you shouldn't have to input these IP Addresses. 2. Call Agent Port The Call Agent port should be 2727. If the Discovery Protocol is working, this option will automatically be set to port 2727.
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MGCP IP PHONES
Troubleshooting
3. Client Port Set this port to 2427. 4. FTP User Name This is the username used to access the Sphere FTP server. The XML decks for the phone's screen are located on this server. 5. FTP Password Password used to access the Sphere FTP server. 6. CTIP Multicast IP The CTIP Multicast IP address should match the multicast IP address configured in the Sphere server software. Please see CTIP Multicast section. 7. CTIP Port The CTIP port should match the port configured in the Sphere server software. 8. Discovery IP The default setting for the Discovery IP is 239.193.0.0. This setting is very important because many of the MGC IP Addresses are automatically set when this is configured. To change the Discovery IP address, please see Changing the Discovery IP Address section. 9. Admin Password The default Admin Password is 22222. This can be changed by performing the following steps from the phone: 1.)Options 2.)9.MGCP Settings 3.)9.Admin Password
Option 10
Phone Status
1. Network Status Shows the network status of the 2 Ethernet ports on the back of the phone. 2. Firmware Status 3. Reset Phone 4. Factory Default Sets the phone back to the factory default settings.
Note: For information on other settings in the options list of the phone, please refer to the Aastra 480i Installer Guide provided with your phone.
2
Browser refers to accessing the phone configuration via a web browser. To access the Aastra 480i Web Client, open your web browser (i.e. Internet Explorer or Netscape, etc) and enter the phone's IP address into the address field, starting with the web prefix http://. In the side menu of the Aastra 480i Web Client, there are 3 main categories: Status, User and Admin. "The Status category contains read only status information for subcategories Network, Hardware and Firmware. "The User category contains user configurable subcategories Reset, Password, and BLF Softkeys. This section is accessed through either the user level or the administrator level user name and password. For more information, refer to the 480i User Guide provided with the phone. "The Admin category contains administrator only configurable subcategories: Network, Firmware and Call Client. This section is accessed through the admin level user name and password.
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MGCP IP PHONES
Troubleshooting
For the administrator, the default user name is "admin" (all lowercase) and password is "22222", and for the user, the default user name is "user" (all lowercase) and password field is left blank.
Verify the phones IP address in the web browser. Click Reset from the Browser window.
Sphericall Managerto Reset the 480i phone from the Sphericall Administrat or
From the Stations tab:
1 2
Highlight the IP phone and right-click and choose Restart. Repeat for any 480i IP phones that need restarted.
Scroll to choose Option 10. Phone Status and press Show softkey, or press 0 to jump directly to this option. Scroll down the Phone Status to 4. Factory Defaults and press Show softkey, or press 4 to jump directly to this option. Press the Set Default softkey to set the phone back to factory default settings. Press Cancel if you wish to exit without resetting to factory defaults.
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MGCP IP PHONES
Troubleshooting
Figure 2.9
STATUS - Hardware
2-34
MGCP IP PHONES
Troubleshooting
Figure 2.11
USER - Reset
2-35
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MGCP IP PHONES
Troubleshooting
2-36
MGCP IP PHONES
Troubleshooting
2-37
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MGCP IP PHONES
...........................................................
General tab. Right-click on System. View Properties. Select the System Initialization Settings tab. Click Add. Select MG Poll Multicast Address from the pull down. Change the default from 239.192.0.0 to your choice.
2-38
MGCP IP PHONES
IP Phone Upgrades
2 3
If prompted, enter the 480i Administrator user name and password and press ok. In the Discovery Multicast IP field, enter the new Discovery IP address. The default is 239.192.0.0.
Press the "Options" button on the phone. Select option "9. MGCP Settings" and enter 480i Administrator password if prompted. Under option "8.Discovery IP", enter the new Discovery IP address. The default is 239.192.0.0.
IP PHONE UPGRADES
...........................................................
All upgrades related to IP phones generate from Sphericall software upgrades. If there is a full system upgrade, IP phones will be upgraded as a part of this process. Refer to the Sphere System Release Notes for full upgrade procedures.
USING
...........................................................
The Quick Reference Guide for this version of IP phone is located in the Appendix of this manual.
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MGCP IP PHONES
Using
2-40
SIP PHONES
...................................
3
Topic
Sphericall supports the SIP standard and are compatible with the Sphere system.
Section Manufacturer Phone
....
Information and overview of SIP parameters Configuring SIP User Agents SIP Failover: Configuring DNS Record for multiple Sphericall Managers
III
Aastra
9112i 9133i 480i 480i CT GXP-2000 GXV-3000 IP30x IP430 IP50x IP550 IP601 IP650 IP4000 F1000G/F3000
IV
Grandstream
Polycom
VI VII VIII
UTStarcom
Installation & Configuration SIP Phone Sphericall Desktop Compatibility Table SIP Phone Star Codes Table
3-41
SIP PHONES
...........................................................
GENERAL SIP CONFIGURATION INFORMATION
Phone Parameters Various models of phones support different configuration parameters. Here are the most common. Outbound Proxy (Cisco, ClearOne, Grandstream, Polycom)The Outbound Proxy field specifies the destination for all SIP requests. If a phone supports both Outbound Proxy and Proxy, Outbound Proxy overrides Proxy as the destination for SIP requests. Most phones allow an IP address, host name or DNS domain name in this field. Proxy (Aastra, Cisco)The Proxy field specifies the destination for SIP requests if the Outbound Proxy is not supported by the phone or is left unspecified. The Proxy field also specifies the host portion of the SIP URI the phone places into the From header of outbound SIP requests and the To header of REGISTER requests. Most phones allow an IP address, host name or DNS domain name in this field. Server/SIP Server (Grandstream, Polycom)Server and SIP Server are aliases for Proxy. Registrar (Aastra)The Registrar field specifies the destination for SIP REGISTER requests. Most phones assume the registrar is collocated with the proxy or outbound proxy. RFC 3263 Locating SIP ServersRFC 3263 specifies procedures that SIP User Agent Clients (UAC) can use to locate SIP servers. Given a domain name and transport protocol, a UAC can locate an ordered list of SIP servers capable of handling SIP requests by using DNS NAPTR and SRV queries. RFC 3263 requires the UAC to send SIP requests to the first server in the list. Should that server be unavailable, the UAC may send the request to subsequent servers in the list until it finds an available server or the list is exhausted. Many phones do not yet support RFC 3263 but some do (Cisco, Polycom). Not all DNS servers support NAPTR records. User Name/User ID (Aastra, Cisco, Grandstream)The User Name field specifies the user portion of the SIP URI the phone places into the From header of outbound SIP requests and the To header of REGISTER requests. Authentication Name/Password (Aastra, Cisco, Grandstream, Polycom)The Authentication Name and Password fields specify the credentials used by the phone to authenticate itself when challenged. Sphericall does not yet challenge SIP phones, so these fields are not currently relevant. SIP Phone MAC Address When a phone checks into a Sphericall system for the first time, the MGC automatically creates a station for the phone. As part of the station creation process, the MGC assigns a unique name to the phone and optionally assigns an extension. For Sphericall MGs and MGCP phones, the MGC uses the phones MAC address as the unique name. For SIP phones, the MGC creates the station when the phone registers with the system for the first time. The MGC uses the SIP URI of the REGISTER request To header as the phones unique name.
3-42
SIP PHONES
As mentioned above, the REGISTER request To header SIP URI is composed of the User Name and Proxy fields. It is important to note that if either of these fields is modified in the phone, the MGC will create a new station and extension for the phone the first time the phone sends a REGISTER request with the new value(s). SIP Domain The SIP Domain (SD) is specified in the SIP Domain field of the System Properties dialog box. When the MGC sends a SIP request to a SIP phone, it populates the host portion of the SIP URI of the From header with the contents of the SD. INVITE, ACK, BYE, CANCEL, OPTIONS and SUBSCRIBE are examples of SIP requests that use the SD in their From header. The SD does not apply to SIP trunks.
RECOMMENDED CONFIGURATIONS
Here are some recommended configurations based upon the capabilities of the SIP phones and DNS infrastructure. IP Address vs. DNS NameDNS host or domain names should be used rather than IP addresses wherever possible for items such as Outbound Proxy and Proxy. Outbound Proxy vs. ProxyOutbound Proxy should be left unspecified if the phone supports both Outbound Proxy and Proxy. Proxy provides greater flexibility since Outbound Proxy specifies the destination for all SIP requests from the phone. For example, the Cisco 7960 is a six line SIP phone. Since each line has a unique Proxy field, its possible to have a line on six different Sphericall systems. If configured with an Outbound Proxy, all six lines must be on the same Sphericall system. SIP DomainThe SD can be an essentially arbitrary string but it should be a valid DNS domain name that can be published on the public Internet. Full RFC 3263 SupportIf the phones and DNS infrastructure support NAPTR queries, the following configuration is recommended: Choose a valid DNS domain name for the SD Example: spherecom.com. Create an NAPTR DNS record for the SD that points to an SRV record for SIP over UDP Example: SIP+D2U _sip._udp.spherecom.com Note that since Sphericall currently only supports UDP, it is not necessary to create records for SIP over TCP or TLS Create an SRV record for SIP over UDP and populate it with the fully qualified domain names of at least two MGCs Example: priority 0, weight 0, voip_manager.spherecom.com Example: priority 10, weight 0, ritas.spherecom.com Set the phones Proxy or Outbound Proxy to the SD Example: spherecom.com. This configuration will allow the phones to locate an alternate MGC should the MGC they are currently registered with fail.
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...........................................................
Sphere has opened the User Agents interface to the system administrator for administration. Required: All SIP endpoints must be listed in this window prior to configuration. Once entered here, the endpoint will be accepted into the full system. AGAIN, this step is required prior to any SIP endpoint being added to the systemtrunk or station. The Sphere system requires this information in order to know how to treat the endpoint or to know what features to apply per endpoint. All widely-used, tested and approved SIP endpoints, for this version of software, are listed in this SIP properties window. All defaults are automatically configured for your convenience. Defaults and supported User Agents are indicated by the check mark. User Agents are also available for adding a new, untested SIP endpoint. If a site has a new untested SIP endpoint, they must add it to the system themselves and verify its operability through their own testing (this testing is not supported by Sphere support personnel). There are generic Agents listed that can suffice for unknown SIP endpoints. Properties of the User Agents can be viewed for appropriate overrides of the default settings for some deployments. All SIP devices of a make, model and firmware version have same attributes, for example, all Polycom IP601 SIP phones running 2.1.0.2708 firmware support "talk" event in the NOTIFY request to answer an incoming call. Instead of assigning these attributes on each SIP endpoint individually, the attributes are assigned to the UserAgent/Firmware-Version binding and gets applied to all corresponding SIP devices. Several new database tables have been added to support this feature. Spheresupported SIP devices (stations and trunks) are pre-configured in the database.
1
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SIP PHONES
Figure 3.1
SIP Properties
2 3 4 5 6 7 8
Review the User Agents listed in the window. If the SIP endpoint you are using is not listed in this window, you must add it. Click Add. Enter the User Agent name. Select Endpoint Type: REQUIRED. The Sphere system requires this information in order to know how to treat it or what features to apply per endpoint. Enter an Agent Description that is appropriate for the Name & Endpoint Type. Click Apply. Note: 1) Those User Agents listed in the SIP dialog window that also have the
Default checked, are those User Agents created into the system by default. These will remain in the system. User Agents added by system administrators are not indicated with a check in the Default column.
Note: 2) If the name of a non-default user agent matches the name of a default user
agent, the Agent Name, Agent Description and Endpoint Type fields cannot be edited. Conversely, if multiple user agent entires exist that have the same name, but none are marked as default, changing any field (other than Version) will change the same corresponding field in the other same-named User Agent entries. The following fields are customizable for entering a non-default user agent:
Table 3.1 User Agent Profile Descriptions
Possible Values
Supported Unsupported (default)
Description
The MGC sends the NOTIFY request to answer/hold/unhold a call remotely (Sphericall Desktop or Web Services).
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Possible Values
Allowed Disallowed (default)
Description
MGC sends a NOTIFY request with terminated (reason=timeout) subscription-state when a SUBSCRIBE request with non-empty to-tag is received and the corresponding subscription is not found. MGC sends a special value of the parameter (Auto Answer, answer-after etc) in the outgoing INVITE request to inform the SIP station to answer immediately. Call Manager creates a SIP phone when not found in the database. Since in 5.2.1+ all Polycom SIP phones are precreated by the system administrator (just like the SIP soft trunks), this capability ensures that these phones are not created by the MGC when not found in the database. When a Polycom phone sends its first REGISTER to check into an MGC, the MGC immediately sends a "503 Service Unavailable" response with a Retry-After timeout of 300 seconds. In the mean time MGC obtains phone configuration from the database. If the configuration from the database is not available, MGC continues sending "503 Service Unavailable" message in response to the REGISTER requests. MGC uses the appropriate method to find the terminal associated with the incoming INVITE request from the endpoint.
Click-To-Dial
Ring Callers Phone First (default) Use answer-after ??param (INVITE:: Call-Info Header) Use auto-answer value (INVITE:: Call-Info Header)
Endpoint Created By
Authentication Info Default (default) DID Mapping From Header URI Outbound Contact URI P-Asserted-Identity Header URI (currently supported)
Hardware Address
SIP has the capability to search for a hardware address in the REGISTER request. (Quintum products only)
MGC sends an unsolicited NOTIFY request when the MWI state changes.
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Possible Values
Supported (default) Unsupported
Description
When the value is set to Supported, the MGC will not send a NOTIFY request to the endpoint even if there is a change in the MWI status. If the value is set to Unsupported MGC will send the unsolicited NOTIFY (out of dialog) request when there is a change in the MWI status even if the endpoint has not sent a SUBSCRIBE request. Setting specifies the maximum packet size Sphericall Media Server should send to the far-end. Setting specifies which SIP endpoints support or do not support OPTIONS request. MGC does not send REFER to initiate a click-to-dial call. MGC sends REFER to initiate a transfer. MGC sends the NOTIFY request to reboot a station. MGC sends the QHeader (Question header in a URI) in the Refer-To header in the REFER request. MGC sends Video SDP in the outgoing INVITE.
OPTIONS Request
Supported Unsupported (default) Supported (default) Unsupported Supported Unsupported (default) Supported (default) Unsupported
Remote Reboot
Video
...........................................................
These tables essentially provide the functionality provided by the MGSetting table, but at a much less overhead. These tables also provide the following benefits: When a Sphericall system is upgraded (pre-6.0 to 6.0+), Sphericall automatically creates UserAgentVersionHubAssociation for all Outbound/None registration type devices. It also creates a binding in UserAgentVersionHubAssociation table for those SIP devices (irrespective of registration type) that have an entry in the deprecated UserAgentHubAssociation table. This ensures that the SIP devices keep functioning after the upgrade. When a Sphericall system is upgraded (6.0 to 6.0+), Sphericall will not change the existing UserAgentVersionHubAssociation bindings. Therefore, the SIP devices overriding the default behavior will continue to work the same way after the
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SIP Failover
upgrade. However, Sphericall may change the default values of the Parameters configured in the Parameter table (this is similar to that of the MGSetting defaults which may infrequently get changed on an upgrade). When the firmware of a SIP device is upgraded, if the SIP device has a forced binding in the UserAgentVersionHubAssociation table, the DbServer/MGC do not update the binding for the new firmware and the SIP device continue the work same way in the MGC. For Outbound/None registration type endpoints DbServer and MGC never update the UserAgentVersionHubAssociation binding even if the SIP device is reporting a completely different User-Agent/Firmware-Version than what is configured in the UserAgentVersionHubAssociation table. A new User-Agent/Firmware-Version can be created and assigned to a custom ParameterProfile and then this User-Agent/Firmware-Version can be bound to the UserAgentVersionHubAssociation and set to "forced" type binding. This feature provides a similar level of control provided by the MGSetting table. SIP station upgradesUpgrades to User Agents with firmware are automatic and apply to all the endpoints on the system. SIP trunk upgradesUpgrades to User Agents are not automatic.
From the Sphericall Administrator application: Open the General System properties. Select the SIP tab. Scroll to view the User Agents. Click the far right column of the User Agent to be removed. Click Remove. Repeat for other User Agents. Click OK to exit.
SIP FAILOVER
...........................................................
CREATE A DNS RECORD
Creating a DNS Record is necessary to determine the Sphericall Manager with which the phones register. Book 1: Plan & Prepare the Sphere System has an extensive discussion of what considerations are important for systems with multiple Sphericall Managers or MGCs.
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SIP Failover
In those systems, in order to provision for failover, it is required for any SIP device to have a DNS record. This affords the phone device a single point of connection, even if the local MGC fails.
Click Start\All Programs\Administrative Tools\DNS. Expand the Forward Lookup Zones tree. Locate your organizations Domain Name. On the Domain Name, right-click New Domain. Type a name to designate the DNS Record. For example, spheresip. Right-click on the newly-created spheresip folder, and select Other New Records. Select Service Location (SRV). Click Create Record.
Figure 3.2
you want phones to check into. Set value to 0 for all Sphericall.
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SIP Failover
If your organization has multiple Sphericall Managers, it is recommended that you establish the following values:
Primary Sphericall Manager
Priority: 0 Weight: 0
f. Port Number: Set value to 5060. The Sphericall Manager uses port 5060, which is the
Click Done. Right-click on the Record name. In this case, spheresip, and select New Host(A). Leave Name field blank. Key in the Sphericall Managers IP address. Click Add Host. Click OK.
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OVERVIEW
...........................................................
Aastras line of SIP phones, with minimal configuration, are interoperable with the Sphere system. This document supports the following Aastra phones: 9112i 9133i 480i 480i CT
Note: For a description of configuration settings specific to each phone, please refer
YOU MUST FIRST VERIFY OR ADD THIS SIP ENDPOINT AS A USER AGENT TO THE SYSTEM. Refer to the beginning of this chapter for this procedure for all SIP endpoints. Plug the phone in and allow the phone to discover an IP address.
a. Select Options b. Arrow down to Option #8 - Network Settings c. Enter the default Admin Password = 22222 d. Arrow down to Option #2 - IP Address
3 4
From a web browser, use the IP address to open a web page for the phone. Select SIP Settings from the lower left-hand side of the web interface and enter the following information: Login: admin Password: 22222
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Overview
Figure 3.3
registrations.
This may be an IP address, host name, or FQDN DNS address. For redundancy and failover, it is recommended that this field be an FQDN DNS address that contains a prioritized list of SRV records. Systems that want to share MGCs across SIP phones can create multiple DNS addresses, each with a different prioritized SRV record list. If you have multiple Sphericall Managers, it is recommended that you use the DNS SRV.
b. Outbound Proxy Port - the MGC uses port 5060, which is the standard port for SIP
traffic.
c. Registrar Server - enter the name/address of the MGC with which the endpoint will
register.
7 8
Test the phone for dial tone. Use the Sphere Administrator application to modify the station properties as appropriate.
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Overview
Figure 3.4
1 2
Select the Explicit MWI Subscription checkbox to Enabled. Enter the requested duration, in seconds, before the MWI subscription times out.
SOFTKEYS/PROGRAMMABLE KEYS
You can configure the softkeys (480i/480i CT) and programmable keys (9112i/9133i) to perform specific functions on the IP phones. The following table provides the number of sofkeys and programmable keys you can configure, and the number of lines available for each type of phone.
Table 3.2 Aastra Phone Operational Features
Programmable Keys Available Handset Keys Available
IP Phone Model
480i 480i CT 9112i 9133i
Softkeys Available
20 20
Lines Available
9 9
15
2 7
1 9
HARDKEYS
There are hard keys on your phone, such as Hold, Redial, Xfer, Icom and Conf (Hold and Icom not available on the 9112i and 9133i) that are configured for specific callhandling features. (See the product-specific User Guide for more information about the hard key functions).
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Overview
You can configure your programmable keys in two ways; either by saving a listing from your Callers, Redial or Directory list to a programmable key through your phone, or by using the Aastra Web user interface.
Press the desired hard key. Scroll through the selected list to find the name and number you wish to save to your speed dial. Press Save. Press the selected speed dial key.
If the selected name is displayed with the number, both are saved to the speed dial.
Access the Aastra Web user interface, according to the instructions above. Select Programmable Keys. Select the Type from the following options:
none speed dial directory line BLF XML Flash services spre park empty pickup lcr
In the Value field, enter a value to associate with the softkey or programmable key.
For the 480i/480i CT and 9133i, in the Line field, select the line for which you want to associate the softkey or programmable key.
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Overview
CONFIGURATION FILES
If your organization intends to deploy hundreds of Aastra phones, the use of configuration files may be more beneficial than the use of the Aastra Web user interface. The Aastra configuration file is supplied on the Sphere CD at Server\Data\Sayson\aastra.cfg A system administrator can enter specific parameters in the configuration files to configure the IP phones. All parameters in configuration files can only be set by an administrator. For a description of each configuration file parameter, refer to the Aastra Administrators Guide at www.Aastra.com.
UPGRADES
Aastra phones can use either TFTP or through the Web interface for upgrades. For use with Sphericall Managers the phones and ease of use administrators will probably want to use the TFTP server.
Note: Sphericall Managers are equipped with TFTP servers.
On reboot, each phone looks for a firmware file in the TFTP directory. If the firmware file exists and the version is different from the installed firmware the Aastra phone performs a TFTP get on the firmware and either performs a upgrade or downgrade depending on the installed firmware version and the firmware version in the TFTP directory. The Aastra phone attempts to locate the following file (located in Program Files\Sphere\Images): 9112i.st 9133i.st 480i.st 480i CT.st
Note: Occurring on 9112i, 9133i and 480i phones, Aastra has documented that their
phones will only reboot if the version of firmware has changed/updated (old or new). Do not expect Aastra phones to reboot immediately after invoking the reboot command.
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Overview
Select the "Firmware Update" tab and enter the IP address of the Sphericall Manager that is running the Sphericall TFTP server. For ease of management, it is probably easiest to have all phones use the same IP address for image downloads, although potentially any phone could be configured with the IP address of any manager if the administrator wants to implement load sharing across the TFTP servers. In order to accomplish an upgrade, the Sphere administrator will be required to reboot each Aastra phone, via the phone keypad, web interface or Sphere Administrator application. Aastra phones reboot only if there is an updated (old or new) firmware is available at the TFTP server. If the firmware is unchanged, or if the TFTP server is not running, the Aastra phone will not restart when MGC sends a NOTIFY request to remoterestart the phone.
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OVERVIEW
...........................................................
The Grandstream GXP-2000 is an enterprise IP telephone based on open industry standards
Note: This product can be utilized with the Sphericall Desktop.
PLANNING
...........................................................
Verify Sphere System Requirements for Grandstream GXP-2000 and interoperability with Sphericall. Verify that the Sphere system is installed, configured and tested as fully functional. Refer to the Grandstream GXP-2000 User Manual for installation planning, setup, package contents, safety and conditions of use.
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Installing
INSTALLING
...........................................................
To integrate the Grandstream GXP-2000 with the Sphere system
Figure 3.7 GXP-2000 Back Panel
POWER
LAN
PC
1 2 3 4 5
YOU MUST FIRST VERIFY OR ADD THIS SIP ENDPOINT AS A USER AGENT TO THE SYSTEM. Refer to the beginning of this chapter for this procedure for all SIP endpoints. Using the supplied power adapter, power the phone up. Connect the phone to the network using the supplied ethernet cable. Allow the phone to discover an IP address. Locate the IP address on the phones LCD display.
Figure 3.8
To locate the IP address, press the circular MENU button to enter menu mode. Use the down arrow button to locate menu item #3 - IP Address.
6 7
Open a web browser. Enter the IP address of the phone into the web browser address field.
Example: //192.168.0.100
8
Enter the default password: admin Note: Only the Administrator can change the password for entry to the Advanced
Settings page. Otherwise, the end user password can be changed on the Basic Settings page.
WEB CONFIGURATION
...........................................................
A series of web pages allow you to configure the GXP-2000s many features.
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Web Configuration
Table 3.3
Required Entries
Information is not configurable from this page. IP Address dynamically assigned via DHCP Speed Dial Configuration Time Zone Configuration No Key Entry Timeout Layer 2 and Layer 3 QOS tagging Local RTP Port TFTP Server Address for Firmware Upgrade SIP Server Outbound Proxy SIP User ID Subscribe for MWI Voice Mail UserID
Advanced Settings
Account 1, 2, 3, and 4
ACCOUNT PAGE
The GXP-2000 has four configurable lines, each uniquely configurable through the Accounts tab on the Device Configuration web page. Lines that are not configured use the Account 1 configuration. Each uniquely defined account appears as a separate device in the Sphericall Administrator application.
Figure 3.9 Account page
will register.
c. SIP User ID: the user name portion of the SIP address for this endpoint. In this case,
the MAC address of this phone. In some instances, the SIP extension is used.
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Web Configuration
There are side effects to filling in certain values under the Account(s) tab: Accounts not configured use the values set in Account 1. The creation of a unique account on the phone requires, at a minimum, a value in the SIP Server field. However, the account and line will not be usable until the SIP User ID is also set. If values are set in an account tab, the minimum information, specified above, is required. The values need not be unique for each account. If the SIP User ID field is entered, but the SIP Server field is not entered, the values from ACCOUNT 1 are used.
10
The phone will appear in the Sphere Administrator application as a SIP phone with a MAC address of <phones mac address>@SIP Server.
11 12
Test the phone for dial tone. Use the Sphere Administrator application to modify the station properties as appropriate.
At this point, the phone will have dial tone, you should be able to place calls, and the device will appear in the Sphere Administrator application. Further configuration is needed to ensure the full functionality of the phone.
Set to Yes.
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Web Configuration
Set to Yes.
SPECIAL FEATURE
This setting enables the third party call control functionality.
Figure 3.15 Account page
Set to Broadsoft.
Click the Basic Settings page. Verify that dynamically assigned via DHCP is selected.
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Web Configuration
Type a name in the Name field which is used to identify the person. It will be displayed on the LCD when pressing the corresponding key. Type a phone number or extension in the UserID filed. Select the appropriate account in the Account field. This is the SIP account associated with the number.
Select the appropriate time zone to control how the date/time is displayed on the LCD.
ADVANCED SETTINGS
QOS
Layer 2 and Layer 3 QoS tagging can be set. The values are applied to RTP and SIP packets, but cannot be set independently for each type of packet. IPv4 provisions two methods for the tagging of packets in the IP header, IPv4 Type of Service (TOS) and Differentiated Services (DiffServ). Although they partition the bits differently, both methods use the same octet in the IP header. Tags are provided merely so that the network infrastructure can prioritize traffic based on the interpretation of these tags. Previously, Spheres System Initialization Settings supported Layer 3 only. Sphericall now supports both Layer 2 and Layer 3 prioritization. These QoS parameters may be set on the QoS tab of the System Properties. Spheres QoS settings are system wide in scope, but please note: any Layer 2 settings that are set on the Sphericall Manager, must also be set at the driver level of the Sphericall Manager, and must also be set on each and every endpoint on the system in order for it to effectively work throughout the system.
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Web Configuration
The Layer 3: QoS field defines the Layer 3 parameter which can be used for IP Precedence or Diff-Serv or MPLS. The default value is 48. The Layer 2: QoS field contains the value used for layer 2 VLAN tagging. The default setting is blank.
Figure 3.19 Advanced Settings page
UPGRADES
Grandstream phones can be configured to use either TFTP or HTTP for upgrades. For use with Sphericall Managers the phone should be configured to use TFTP.
For ease of management, it is probably easiest to have all phones use the same IP address for image downloads, although potentially any phone could be configured
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SIP PHONES
Web Configuration
with the IP address of any manager if the administrator wants to implement load sharing across the TFTP servers. The Grandstream GXP-2000 attempts to locate the following files (located in Program Files\Sphere\Images): boot55.bin gxp2000.bin ring1.bin ring2.bin ring3.bin If the phone is unable to find the TFTP server or one of the image files, the phone aborts the TFTP transfer and simply uses the image stored in flash. The phones will take slightly longer to boot when image files are located in the TFTP server directory. Depending on the connection speed and size of the image file, Grandstream states that the boot process may take up to 5 minutes. If the boot times becomes an issue the administrator may remove the files from the Images directory. However, if the files are removed any new Grandstream phones added to the system may not boot up with the image shipped and compatible with the Sphere system. In order to accomplish an upgrade. the Sphere administrator is required to reboot each Grandstream phone either via the web interface, phone keypad or, power cycle.
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OVERVIEW
...........................................................
The Grandstream GXV-3000 is a next generation enterprise IP telephone based on open industry standards.
Note: This product is intended to operate as an independent device, and therefore,
PLANNING
...........................................................
Verify Sphere System Requirements for Grandstream GXV-3000 and interoperability with Sphericall. Verify that the Sphere system is installed, configured and tested as fully functional. Refer to the Grandstream GXV-3000 User Manual for installation planning, setup, package contents, safety and conditions of use.
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Installing
INSTALLING
...........................................................
To integrate the Grandstream GXV-3000 with the Sphere system
Figure 3.23 GXV-3000 Back Panel
1 2 3 4 5 6 7
YOU MUST FIRST VERIFY OR ADD THIS SIP ENDPOINT AS A USER AGENT TO THE SYSTEM. Refer to the beginning of this chapter for this procedure for all SIP endpoints. Using the supplied power adapter, power the phone up. Connect the phone to the network using the supplied ethernet cable. Allow the phone to discover an IP address. Locate the IP address on the phones video display. Open a web browser. Enter the IP address of the phone into the web browser address field.
Example: //192.168.0.100
8
Enter the default password: admin Note: Only the Administrator can change the password for entry to the Advanced
Settings page. Otherwise, the end user password can be changed on the Basic Settings page.
WEB CONFIGURATION
...........................................................
A series of web pages allow you to configure the GXP-3000s many features.
Table 3.4 Grandstream GXP-3000 Web Device Configuration table
Web Page
Status
Required Entries
Information is not configurable from this page.
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Web Configuration
Web Page
Basic Settings
Required Entries
IP Address dynamically assigned via DHCP Speed Dial Configuration Time Zone Configuration No Key Entry Timeout Layer 2 and Layer 3 QOS tagging Local RTP Port TFTP Server Address for Firmware Upgrade SIP Server Outbound Proxy SIP User ID Subscribe for MWI Voice Mail UserID
Advanced Settings
Account 1, 2, 3, and 4
ACCOUNT PAGE
The GXP-3000 has four configurable lines, each uniquely configurable through the Accounts tab on the Device Configuration web page. Lines that are not configured use the Account 1 configuration. Each uniquely defined account appears as a separate device in the Sphericall Administrator application.
Figure 3.24 Account page
will register.
c. SIP User ID: the user name portion of the SIP address for this endpoint. In this case,
the MAC address of this phone. In some instances, the SIP extension is used.
There are side effects to filling in certain values under the Account(s) tab: Accounts not configured use the values set in Account 1.
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SIP PHONES
Web Configuration
The creation of a unique account on the phone requires, at a minimum, a value in the SIP Server field. However, the account and line will not be usable until the SIP User ID is also set. If values are set in an account tab, the minimum information, specified above, is required. The values need not be unique for each account. If the SIP User ID field is entered, but the SIP Server field is not entered, the values from ACCOUNT 1 are used.
10
The phone will appear in the Sphere Administrator application as a SIP phone with a MAC address of <phones mac address>@SIP Server.
11 12
Test the phone for dial tone. Use the Sphere Administrator application to modify the station properties as appropriate.
At this point, the phone will have dial tone, you should be able to place calls, and the device will appear in the Sphere Administrator application. Further configuration is needed to ensure the full functionality of the phone.
Set to Yes.
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Web Configuration
Set to Yes.
SPECIAL FEATURE
This setting enables the third party call control functionality.
Figure 3.30 Account page
Set to Broadsoft.
Click the Basic Settings page. Verify that dynamically assigned via DHCP is selected.
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SIP PHONES
Web Configuration
Type a name in the Name field which is used to identify the person. It will be displayed on the LCD when pressing the corresponding key. Type a phone number or extension in the UserID filed. Select the appropriate account in the Account field. This is the SIP account associated with the number.
Select the appropriate time zone to control how the date/time is displayed on the LCD.
ADVANCED SETTINGS
QOS
Layer 2 and Layer 3 QoS tagging can be set. The values are applied to RTP and SIP packets, but cannot be set independently for each type of packet. IPv4 provisions two methods for the tagging of packets in the IP header, IPv4 Type of Service (TOS) and Differentiated Services (DiffServ). Although they partition the bits differently, both methods use the same octet in the IP header. Tags are provided merely so that the network infrastructure can prioritize traffic based on the interpretation of these tags. Previously, Spheres System Initialization Settings supported Layer 3 only. Sphericall now supports both Layer 2 and Layer 3 prioritization. These QoS parameters may be set on the QoS tab of the System Properties. Spheres QoS settings are system wide in scope, but please note: any Layer 2 settings that are set on the Sphericall Manager, must also be set at the driver level of the Sphericall Manager, and must also be set on each and every endpoint on the system in order for it to effectively work throughout the system. The Layer 3: QoS field defines the Layer 3 parameter which can be used for IP Precedence or Diff-Serv or MPLS. The default value is 48. The Layer 2: QoS field contains the value used for layer 2 VLAN tagging. The default setting is blank.
Figure 3.34 Advanced Settings page
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Web Configuration
UPGRADES
Grandstream phones can be configured to use either TFTP or HTTP for upgrades. For use with Sphericall Managers the phone should be configured to use TFTP.
For ease of management, it is probably easiest to have all phones use the same IP address for image downloads, although potentially any phone could be configured with the IP address of any manager if the administrator wants to implement load sharing across the TFTP servers. The Grandstream GXV-3000 attempts to locate the following files (located in Program Files\Sphere\Images): boot64.bin load64.bin gxv3000a.bin ring1.bin ring2.bin ring3.bin If the phone is unable to find the TFTP server or one of the image files, the phone aborts the TFTP transfer and simply uses the image stored in flash. The phones will take slightly longer to boot when image files are located in the TFTP server directory. Depending on the connection speed and size of the image file, Grandstream states
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Web Configuration
that the boot process may take up to 5 minutes. If the boot times becomes an issue the administrator may remove the files from the Images directory. However, if the files are removed any new Grandstream phones added to the system may not boot up with the image shipped and compatible with the Sphere system. In order to accomplish an upgrade. the Sphere administrator is required to reboot each Grandstream phone either via the web interface, phone keypad or, power cycle.
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...........................................................
Please verify Sphericall and Polycom version compatibility in the Sphere System Requirements document. Verify with the System Requirements document the compatability of Polycom phones (IP30x, IP430, IP50x, IP550, IP601, IP650, and the IP4000). The following documentation can be applied to any of these SIP phones.
PLANNING
...........................................................
Verify Sphere System Requirements for the appropriate Polycom phone and interoperability with Sphericall. If you deploy SIP2.2.0 software release for Polycom or higher, the 300 and 500 will not have feature support. Most 300 and 500 phones are running MGCP, but those who may be converting from MGCP to SIP could run into this problem if/when the phones are flipped to SIP. Sphere system should be installed, configured and tested as fully functional. Refer to the Polycom User Manual for installation planning, setup, package contents, safety and conditions of use. Verify the Polycom phone keycap labels that are appropriate for this application. Customization of microbrowser information on some units can be made by developers. Sphere has a document available for developers in a Software Development Kit (SDK) that specifically addresses support for microbrowsers. The Polycom SIP IP650 can support the wide band CODEC G.722. G722 should be higher priority than PCMU in both the LAN and WAN preferred CODEC lists.
Note: This product is intended to operate as an independent device, and therefore,
OVERVIEW
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Sphericall IP phone support is being divided into tiers. For tier 1 phones, Sphericall will support native features of the phone. In addition, Sphericall will develop tools that aid in the commissioning, configuration and maintenance of the phone. Each tier 1 phone family will have their own test plan created specifically to test features supported by the phone. Tier 2 phones are not supported to the same depth as tier 1. Tier 2 phones are functionally tested in a Sphericall environment to verify compatibility. Tools for commissioning and configuration are supplied by the phone vendor and are not integrated within Sphericall.
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Overview
This document describes the commissioning, configuration and maintenance of the tier 1 Polycom SIP phone family which includes the IP320, IP330, IP430, IP501, IP550, IP601, IP650, and the IP4000.
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Overview
Polycom has two versions of this file: one for SIP and the other for MGCP. The Sphericall MGCP implementation currently relies on this file being the MGCP version. Since each Polycom SIP phone must have a unique registration file, the 000000000000.cfg file will be left as the MGCP version. The Polycom SIP version of the 000000000000.cfg file is being renamed to 000000000000_sip.cfg so it is not confused with the MGCP version. <MAC>.cfg The <MAC>.cfg file (e.g. 0004f205b4f6.cfg) is the master configuration file for a specific phone. If a MAC-specific master configuration file does not exist, the phone will download the 000000000000.cfg file. The <MAC>.cfg file contains a list of other files (application image and configuration files). No configuration parameters are to be entered into this file. The <MAC>.cfg file is created by first using the SIP version of the 000000000000.cfg file (000000000000_sip.cfg) as a template. The file list is then updated to include configuration files that override some of the parameters from the default files. E.g.
<?xml version="1.0" standalone="yes"?> <!-- Template for Per SIP Phone Master Configuration Files--> <!-- Edit and rename this file to <Ethernet-address>.cfg for each phone.--> <!--Last updated: 21-February-2007 16:55:58 Vyskocil, Randy--> <APPLICATION APP_FILE_PATH="sip.ld" CONFIG_FILES="phone_cust.cfg, 0004F211CE9E_phone_overrides.cfg, Spanish_Spain.cfg, phone1.cfg, sip_overrides.cfg, sip.cfg" MISC_FILES="" LOG_FILE_DIRECTORY="" OVERRIDES_DIRECTORY="" CONTACTS_DIRECTORY=""/>
The order of the file list is important. The IP phone parses the file from the beginning of a line to the end. The first occurrence of a parameter is used, thereby, overriding any other occurrence of the same parameter. This file and the <MAC>_phone_overrides.cfg file can be recreated from information stored in the configuration database. sip.cfg The sip.cfg file contains basic operation parameters that are common to all Polycom SIP phones. This file is provided by Polycom and will not be modified by Sphere. sip_overrides.cfg The sip_overrides.cfg file is a Sphericall-created file that contains sip.cfg override parameters. This file has parameters that may either be administratively controlled, defaulted to something different than the values in sip.cfg or both. The parameters in sip_overrides.cfg that are administratively controlled are also stored in the Sphericall configuration database. <language_country>.cfg The <language_country>.cfg file is a Sphericall-provided file that contains call progress parameters unique for a specific country. The <language_country> portion of the name will correspond to the sub-directory names of the Polycom SoundPointIPLocalization folder.
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Overview
phone1.cfg The phone1.cfg file is a Polycom-provided template that contains parameters unique for a particular phone. This file will not be modified by Sphere. phone_overrides.cfg The phone_overrides.cfg file is a template file from which <MAC>_phone_overrides.cfg files are created. Phone_overrides.cfg contains all the phone-specific parameters that can be manipulated using Sphericall via the <MAC>_phone_overrides.cfg files. This file contains elements of both sip.cfg and phone1.cfg parameters. <MAC>_phone_overrides.cfg The <MAC>_phone_overrides.cfg file contains all the parameters for a particular phone that can be manipulated using Sphericall. The parameters in <MAC>_phone_overrides.cfg are also stored in the Sphericall configuration database. phone_cust.cfg The phone_cust.cfg is the repository for settings made by the customer that should be applied to all Polycom SIP phones. Except for an XML header, this file will be empty on new installations. Sphericall upgrades will not modify this file. A reference to phone_cust.cfg will be placed in all <MAC>.cfg files in such a way as to override parameter definitions in other configuration files. <MAC>_phone_cust.cfg The <MAC>_phone_cust.cfg is the repository for settings made by the customer for a particular phone. Since the majority of the time customer overrides won't be used/ needed, this file is only created and deleted on-demand. This file's existence and contents can be determined from the Sphericall Administrator application. <MAC>-phone.cfg The <MAC>-phone.cfg is created by the phone and uploaded to the server whenever the user makes a configuration change using the phone's menu key or web interface. Changes made via the phone or web interface override all other matching setting within the configuration files. Past Sphericall implementations for configuring the Polycom MGCP phone manipulated this file. The Polycom phone provides a facility ("Reset to Default") through its Menu key to reset user changes. The Sphericall Administrator application displays the contents of this file. It also provides a command for deleting the file's contents. Note: When upgrading a Polycom SIP phone from v5.1 to v6.0, each phonespecific file must be deleted before the upgrade. Once this is done, the phone can be configured in the Sphericall Administrator application (in Sphericall v6.0).
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Overview
000000000000-directory.cfg The 000000000000-directory.cfg file contains the default local contact directory. If a <MAC>-directory.cfg doesn't exist the phone will use this file. This is an optional file and will be managed by the customer. <MAC>-directory.cfg The <MAC>-directory.cfg file contains the local contact directory for a specific phone. Polycom provides a "Directory" button to access and edit local contacts. Changes made by the phone can be used immediately and saved to this file without requiring a reboot. If this file is changed outside of the phone, the phone must be rebooted to load the changed file. This is an optional file and will be managed by the customer.
SYSTEM OVERRIDES
System-wide parameters that can be managed by Sphericall are stored in sip_overrides.cfg. This file also contain parameters meant to override Polycom defaults, but are not managed by Sphericall.
DIGIT MAP
Digit maps are used to eliminate the need to press the send button on the Polycom SIP phone. A digit map override allows extension dialing without having to wait for a digit timeout. Polycom phones follow RFC 3435 "Media Gateway Control Protocol (MGCP)" for its digit map syntax. The default digit map assumes the following:: The North American PSTN numbering plan A minimum of 3 digit extensions A PSTN access digit of '8' That 0 is a valid extension The digit map can be accessed/overridden from the phone_cust.cfg file. The default digit map for Polycom SIP phones is: digitmap dialplan.digitmap="8[29]11|80T|8011xxx.T|8[0-1][2-9]xxxxxxxxx|8[2-9]xxxxxxxxx|0|[17]xx.T|9xx.T|*7[246]xxx.T|*7[3789]|*75x.#x.T|*8x|*91x|*9[25]|*9[346]xxx.T|*971[01]|*9 9x.#xxx.T" The default digit map segments have the following meaning:
8[2-9]11 80T 8011xxx.T 8[0-1][2-9]xxxxxxxxx 8[2-9]xxxxxxxxx 0 [1-7]xx.T Outside sevice -X11 e.g. 411, 911 Outside operator International calls Long distance calls 10 digit dialing Inside operator Extensions - Minimum 3 digits
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Overview
|9xx.T|
If 911 is setup as an emergency number for the telephony area of the phone then dialing 911 will progress as an outbound emergency call. If internal numbers can start with a 9 then this rule allows 3 or more digit extensions to be dialed. Star code - forwarding, transfer to VM, intercom Star code - deactivate forwarding, drop last call, DND on, DND off Star code - assign account code Star code - park at zone Star code - unpark at zone Star code - group pickup, conference Star code - pickup at extension, blind transfer Star code - MWI on, MWI off Star code - user authentication
Refer to Polycom Technical Bulletin 11572 for dial plan configuration information. Any changes in the digit map will require all affected Polycom SIP phones to be rebooted. The digit map defaults can be tailored per country and stored in the <language_country>.cfg files. Polycom doesn't provide any support.
QOS PARAMETERS
802.1p/Q Ethernet user priority settings and IP Type of Service (TOS) settings are supported. These settings will control both Polycom SIP and MGCP endpoints.
LOCALIZATION
LANGUAGE
Polycom phones can be configured to display other languages besides English. The implementation is extensible where other languages can be added by third parties. Sixteen different languages are currently being shipped which include: English / US, English/UK, French/France, German/Germany, Italian/Italy, and Spanish/Spain. The other languages that Sphere is interested in, but not currently supported, are French/ Canadian and Spanish/Mexico. The bootrom messages are not localized and are displayed in English. The language of the default localization setting is set as the language for each new phone.
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Overview
The system override for Polycom SIP phone language will be configured via the Sphericall Administrator applications Localization Settings.
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Overview
Using the supplied power adapter, power the phone up. Connect the phone to the network using the supplied ethernet cable.
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Overview
The MAC Address can be filled in using the keyboard or bar code reader. It is limited to 12 hexadecimal characters.
7
Enter the extension that will become primary for this phone in the Extension field. If necessary, click Add to create a new extension for this phone.
Select from the available, supported Polycom SIP phones in the drop down box.
The phone type choices are derived from the supported user agent strings. Any supported user agent that contains "Polycom" will be listed. If you select an incorrect Polycom phone, the system corrects your choice.
9
Click OK.
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Overview
Updated in <MAC>_phone_cust.cfg
This is auto-generated from the stations primary address and the systems DNS domain.
12
Type the text that appears next to the line key on the phone.
This label is limited to 12 characters and defaults to the primary extension number.
13
Type the line key for the number of phone keys that should be bound to the phones registration.
If Line Keys is set to greater than 1, the phone will be set to limit the calls per line to 1. In the Server area:
14
Enter the DNS address of one or more MGCs that accept SIP registrations.
This may be an IP address, host name, or FQDN DNS address. For redundancy and failover, it is recommended that this field be an FQDN DNS address that contains a prioritized list of SRV records. Systems that want to share MGCs across SIP phones can create multiple DNS addresses, each with a different prioritized SRV record list. If you have multiple Sphericall Managers, it is recommended that you use the DNS SRV.
15
Select the appropriate transport settings in the Transport and Port fields
This is the interaction between these fields and the server address.
Note: Refer to the Polycom administration guide for settings. Note: Do not enter a setting in the Port field if NAPTR or SRV records are used.
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Overview
Parameters that are set at the phone or using the Polycom web interface are saved in an XML configuration file (<MAC>-phone.cfg). The Phone Managed Override Parameters field displays the contents of this file, if it exists. Pressing the Reset button will clear the contents of this file. In the Custom File Parameters area:
17
This field displays the contents of the <MAC>_phone_cust.cfg file, if it exists. The file may also be deleted by pressing the Delete File button. In the VM Address field:
18
Enter the address for contact when the Voice Mail Messages button is pressed.
The Idle Screen URL is the address the phones micro-browser periodically contacts to update what is on the idle display. The Refresh Rate (in seconds) is used to determine how often the idle display is updated. The Refresh Rate defaults to 15 minutes (900 seconds). In the Services URL field:
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The phone changes its display to act as a browser, using the arrow keys to navigate.
21
Press the DNS Test button to verify the DNS address of the Server and Outbound Proxy.
DNS Test will query DNS resolving NAPTR records. The DNS "Server Address" and "Outbound Proxy Address" will always be checked for A records and SRV records (records along with priorities and weights), as well as NAPTR records. It will be up to the administrator to interpret the results and what affect, if any, it has on the Polycom SIP phone. The DNS NAPTR query is not attempted when the Sphericall Administrator application is run under Vista.
22
Click OK.
A warning message window will remind you to bind a user to the primary extension when a Polycom SIP phone is created.
Copy/paste <MAC>directory.xml and rename as <the phones MAC address>-directory.xml. Open the file in Notepad and make modifications to the fields described in the following table.
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Overview
Table 3.5
Permitted Values
UTF-8 encoded string of up to 40 bytes UTF-8 encoded string of up to 40 bytes UTF-8 encoded string containing 8 digits (the user part of a SIP URL) or a string that constitutes a valid SIP URL
Interpretation
first name last name contact Cannot be null or duplicated; is used by the phone to address a remote party in the same way that a string of digits or a SIP URL are dialed manually by the user. This element is also used to associate incoming callers with a particular directory entry. speed-dial index Associates a particular entry with a speed dial bin for onetouch dialing or dialing from the speed dial menu.
sd
Null, 1 to 9999
rt
Null, 1 to 21
ring type When incoming calls can be associated with a directory entry by matching the address fields, this field is used to specify ring type to be used.
dc
UTF-8 encoded string containing 8 digits (the user part of a SIP URL) or a string that constitutes a valid SIP URL
divert contact When incoming calls can be associated with a directory entry by matching the address fields, this field is used auto divert If 1, automatically diverts callers that match the directory entry to the address specified in divert contact.
ad
0,1
ar
0,1
auto reject If 1, automatically rejects callers that match the directory entry.
bw
0,1
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Overview
Element
bb 0,1
Permitted Values
Interpretation
buddyblock If 1, block this contact from watching this phone.
From the Sphericall Administrator application. Click on File/Export. Select Phone Distribution Map from the Export dropdown field.
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Select the fields of information you require or this report. Select Export. Enter the name of the file and location for the .csv file that will be generated. Save. Open the file using Excel.
Optimally the report would provide First Name and Last Name and possibly Username along with the MAC address and primary address. The MAC address will appear in the Phone S/N (Mac) field.
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Overview
UPGRADES
BACKUP
During a Sphericall upgrade, the Polycom root directory is backed up into "backup" directory.
FTP SERVICE
The Polycom phones require a local user account on the FTP server to have access to images and configuration files. The default FTP user account name is "PlcmSpIp" with a password of "PlcmSpIp. If the FTP server resides on a Microsoft server, the password will not be compliant with Microsoft's default password complexity rules. To help in debugging phone issues, a single file directory is used to hold images, configuration and log files.
LOG FILES
Polycom SIP phones will create or rewrite a boot log (e.g. 0004f21114e4-boot.log) whenever the phone is restarted. Boot log files are useful when debugging image and configuration file issues.
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Overview
Polycom also provides uploading of application log files (e.g. 0004f21114e4-app.log). Empirical evidence has indicated the phone does not create the application file but it must exist before the phone will write to it. During commissioning of the phone, the Sphericall Administrator will create a blank application log file if optioned to do so. Application log files are useful in debugging content issues within configuration files. The FTP server must be configured to allow write access for log files to be uploaded.
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OVERVIEW
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The F1000/F3000 Wi-Fi handset is a device that expands the Sphere system over the enterprise wireless IP network for enhanced mobility of IP communications. Consider the F1000s potential as an effective way to communicate by allowing mobile users, within the enterprise, access to the IP PBX functionality. For example, the flexibility to take the phone out on the shop floor. The F1000/3000 includes the following features: Caller ID Blind Transfer Attended Transfer 3 Party Conferencing Sphericall Voice Mail Access
Note: The G in the UTStarcom F1000/3000 represents an 802.11g radio. All other
aspects of the phones form, fit and function remains the same as the F1000/3000.
THINGS TO CONSIDER
Before attempting to install and configure a UTStarcom F1000/3000 phone, you will need to consider the following: Do you have rights to the DNS Server? Are you familiar with your organizations wireless security standards?
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SIP PHONES
Overview
Do you have Admin rights to access the Sphericall Administrator application on the Sphericall Manager?
Note: This product is intended to operate as an independent device, and therefore,
can not be utilized with the Sphericall Desktop. This document is divided into three sections. Each is necessary for successful installation and configuration of the F1000/3000 phone.
Create a DNS Record on the DNS server A. B. C. D. Configure WiFi and network settings from each F1000/3000 phone Access and configure phone settings from a web interface Configure initialization settings for F1000/3000 and Sphericall Voice Mail Firmware Updates
Press the Menu key. Using the WiFi down arrow key, scroll to WiFi settings. Click OK. Go to WiFi Config and click OK. Select AP Profile. SSID1 is one of the default profiles associated with the F1000/3000. You will need to rename this to the name of the wireless network access point with which the F1000/3000 will connect.
Your organization may have multiple wireless network access points. You may need to add multiple profiles to ensure phone connectivity throughout the network.
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Select SSID1 and click Edit. Click OK. From the phone keypad, enter the name of the appropriate wireless network access point. Click OK. Scroll to Security Mode and click OK. Select WEP Key and click OK.
Short for Wired Equivalent Privacy, a security protocol for wireless local area networks (WLANs) defined in the 802.11b/g standard. WEP is designed to provide the same level of security as that of a wired LAN. WEP aims to provide security by encrypting data over radio waves so that it is protected as it is transmitted from one end point to another. Your organizations wireless security standards may not require WEP.
12
Select the appropriate security mode for your organization and click OK.
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Overview
13
Your wireless network administrator may need to supply you with the WEP KEY password.
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Click OK on the Activate menu. After completing WiFi settings, switch off the phone and then switch it on, in order for the changed WiFi settings to take effect. To retrieve the F1000/3000s IP Address, press Menu\WiFi Settings\Network Info.
Make note of the phones IP address, as you will need it for web configuration.
From a web browser, use the F1000/3000s IP address to open a web page for the phone. At the prompt, enter the following information:
a. User Name: user b. Password: 888888
From the navigation pane on the left side, select the User Menu\SIP and RTP Config option.
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Overview
DNS Record.
k. Set Registration Duration: Set value to 3600 seconds. Standard SIP values range
Click Submit. Click Reboot. Note: Upon reboot, the F1000/3000 phone will initialize and the phone will appear in
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Overview
From the navigation pane on the left side, select the User Menu\User Settings option. Key in the following user information:
a. Display Name: Type in the name as you want it displayed on the LCD of the
F1000/3000.
b. Autoscan Flag: Scans the wireless network for seamless transition from one network
Click Submit.
From the navigation pane on the left side, select the User Menu\Wireless AP Config option. Make the necessary changes to wireless settings. Click Submit.
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Overview
YOU MUST FIRST VERIFY OR ADD THIS SIP ENDPOINT AS A USER AGENT TO THE SYSTEM. Refer to the beginning of this chapter for this procedure for all SIP endpoints. Click F5 or select File\Refresh to make the recently-installed F1000/3000 viewable on the Sphericall Administrator application. Click the Stations tab. Highlight the F1000/3000s station.
Hint: If you use the Sort By: Device Type feature, the phone will reside under the SIP Phones folder.
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Right-click on the F1000/3000 and select View Properties. Click the Settings tab. Click Add. Select RTP receive packet size maximum from Sphericall Media Server. Set value to 20. Click OK.
D. FIRMWARE UPDATES
Note: Refer to the System Requirements documentation and the Software
Compatibility table to ensure the F1000/3000s firmware version is compatible with the Sphere system.
Open the following folder from the Sphericall software CD: \Server\Data\UTStarcom
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Overview
2 3
Unzip the F1000/3000 TFTP Package.zip file. Run fwupgrade.exe to push the latest image down to each individual phone. Note: This executable must be run for each F1000/3000 phone that requires an
update.
Figure 3.46 F1000/3000 Upgrade Utility
4 5
Enter the IP Address and click UpLoad. When the utility completes the upload process, the F1000/3000 should reboot and recognize the new image.
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Section VII - SIP Phone Compatibility and Capability with Spherciall Desktop
SECTION VII - SIP PHONE COMPATIBILITY AND CAPABILITY WITH SPHERCIALL DESKTOP
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The following SIP phones have features that are compatible with the Sphericall Desktop.
Table 3.6
Desktop Feature
Controlled by Sphericall Desktop Click-to-Dial Click-to-Answer Transfer (Blind: connected and ringing) Transfer (Attended) Hold Pick-up Park Do Not Disturb Three Way Call Call Recording Drag-n-drop MeetingHub Conference (blind transfer to a conference extension) Precedence Dialing (CallNOW/MLPP) Phonebook Directory Recent Calls List Dual Mode (Softphone and Computer Telephony) Media Switching (PC to Phone) Video Calling Instant Messaging - Send & Receive
Yes Yes Yes Yes Yes6 Yes Yes Yes Yes5 No4 Yes Yes
Yes Yes Yes Yes Yes6 Yes Yes Yes Yes5 No4 Yes Yes
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Desktop Feature
Presence Agent Manage Forwarding Profiles Star Code Support PIN Code Support
1
Yes3 Yes No No
Yes Requires that IM be disabled on the IP phone Yes2 Requires that IM be disabled on the IP phone Yes3 Requires that Presence be disabled on the IP phone Yes4 Note: MLPP is not supported on the SIP phones.Precedence receiving is not possible using a SIP phone Yes5However, the local DND indicator will not be reflected on the IP phone. If DND is set at the phone, the Sphericall Desktop will not reflect this. Yes6However, the first call must be placed on hold before initiating the consult call. Simply dialing the second call will not currently work. No1When the Sphericall Desktop is associated with a SIP phone, it can not be used as a softphone to terminate media. No2Due to No1, media cannot be terminated at the PC. No3SIP phones require that media only be terminated in one place (video and audio together). No4Three way conferences can not be established by the Sphericall Desktop that is associated with a SIP phone.
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Administrative Star Codes
Sphericall Star Code
*74 + extension
Explanation
Allows users the ability to forward phone calls directly to another users voice mail, thus bypassing the ringing of the users extension. The Sphericall Manager sends call detail information to the call detail record.
Callers without intercom can enter this star code to intercom a phone equipped with intercom.
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SIP PHONES
Explanation
Allows administrators to assign an extension number to a station within a Sphere system. Use the *97 administrative star code to confirm the stations assigned extension number. Note: A confirmation tone will sound after applying this star code command.
Table 3.8
Diagnostic Action Forcibily turn MWI lamp ON Forcibily turn MWI lamp OFF
Explanation
This turns on the message waiting indicator on the phone set (analog & IP phones with MWI lamp) This turns off the message waiting indicator on the phone set (analog & IP phones with MWI lamp) Reserved for future use.
*9710
*973-*979
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Session Initiation Protocol, SIP, is a protocol for transporting call setup, routing, authentication and other feature messages to endpoints within the IP domain. Within the Sphere system, SIP is used to allow external systems to participate in calls with the Sphericall Manager. The Manager targets the use of SIP for integration with some specific third-party products for integration with the following: two-way calls (Sphericall Manager and external system), calls between two systems placed on hold, transfer of calls between the two systems, passing of DTMF digits into the thirdparty system during a call, and notification of message waiting. The Terminal location logic to accommodate gateways that register multiple trunks from the same IP address. Understanding this logic will help enormously when troubleshooting SIP connection issues. The Sphericall Manager (MGC) uses the following logic to locate a SIP terminal in 6.0. and later: From Header Userinfo is compared to the Account field of the Service Provider information. If no match is found, the Sphericall Manager moves to step 2.
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Note: this is the most common way stations are identified, but does not help for trunks since the FROM field contains the caller ID of the incoming call. To Header Userinfo is compared to the DID maps configured for SIP trunks. If two trunks have overlapping DID maps, the Sphericall Manager moves to step 4. If no match is found, the Sphericall Manager moves to step 3. Contact Header hostname:port is compared to the Outbound Proxy if configured, otherwise the Service Provider Domain of the Service provider information. If the hostname:port is an IP address it is compared exactly to what is configured. If the hostname:port is not an IP address, a partial compare is performed against the Service Provider information. For example, the hostname "horatio.rndlab.spherecom.com" would match the Service Provider information "rndlab.spherecom.com". Note: in both cases the port must match. If more than one User Agent matches this criterion, or no match is found, the Sphericall Manager moves to step 4. Authorization Header If the request contains an Authorization header, the credentials included in the Authorization Header are compared against the credentials configured in the Sphericall Admin Authorization window. If no match is found, the Sphericall Manager moves to step 5. The Sphericall Manager (MGC) challenges the sender to obtain credentials via the Authorization Header.
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PLANNING
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Verify Sphere System Requirements for RG613 and interoperability with Sphericall. Verify the required firmware version in the Sphere System Requirements. The RG configuration is also compatible with the iMG. Wherever you see RG in these instructions (except for command line interface entries) you may substitution iMG.
Note: ATI manufactured serial cable is required for configuration:
Model number: AT-RGCONSOLECABLE-00 Part number: 990-11748-00 Sphere system should be installed, configured and tested as fully functional. Refer to the AT-RG613/623TX Residential VoIP Gateway User Manual for installation planning, setup, package contents, safety and conditions of use.
PREPARING
Figure 4.1 RG613 Back Panel
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Voice Ports TEL 2 TEL 1
10BASE-T/100BASE-TX NETWORK PORTS LAN 3 LAN 2 LAN 1 WAN 1
POWER
Along with ordering RG613, you must order at least ONE of the Allied Telesyn proprietary serial cables for coding the residential gateway:
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AT I R G 6 X X O R I M G 6 X X
Installing
Prepare network according to Sphere System Requirements. Verify that a SNTP (Simple Network Time Protocol) time server is configured. Know the required login for working with this gateway: Login: manager Password: friend Connect appropriate network cable to appropriate port on RG613.
INSTALLING
To verify firmware version of RG or iMG
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The RG configuration is also compatible with the iMG. Wherever you see RG in these instructions (except for command line interface entries) you may substitution iMG.
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Connect serial cable on RG613 to lap top computer for configuring device. Connect power cable to the RG613 unit. Verify it powers up. Connect all network connections, there will be a firmware check by the unit later. Open a HyperTerminal Session:
Serial Port Info: Bps: 38400 Data: 8 Parity: None Stop Bits: 1 Flow Control: None
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Verify that HyperTerminal has an open session. Power the unit off and back on again.
You will now see the RG boot process and the firmware will check in.
7
Verify the MG version using HyperTerminal. Note: If the version verified is v1.0, you will need to upgrade the Flash. If the version
verified is v2-3 or greater, you may proceed to RG configuration (please verify the current supported version of firmware with the Sphere System Requirements).
To update f lash of RG
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Login: manager Password: friend --> ip set interface ip0 dhcp enabled --> dhcpclient update --> system config create <filename>.cfg
--> system config set <filename>.cfg (as created above) --> console enable ati [enter] [enter]
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AT I R G 6 X X O R I M G 6 X X
Installing
8 9
[RG ipaddress]\ tftp> connect [ip address of sphericall manager to connect to] getflash rg6xx-image-2-2_06-2-4_58.bin (This is an example of the syntax. Verify the current flash image required for this in the Sphere System Requirements)
Wait for the RG to get the image and reboot itself (3-5) minutes. NOTE: do NOT reboot the RG manually at this time. This will corrupt the flash. After the RG has rebooted:
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Do not close HyperTerminal at this time--it must remain open. Open the following folder on the Sphericall DVD: Server\Data\AlliedTelesyn Run the Loader_RG600_2-5_55_10_04.exe to push the latest image down to the RG or iMG (This is an example of the syntax. Verify the current
When the window opens during this process, enter the IP address you retrieved from the ip list interfaces step (above). Enter password: friend. Click Start. Watch the HyperTerminal window to verify the reboot of the RG. This indicates that the image download is complete (check Sphere System Requirements for current version number). Continue with RG configuration as follows.
17
To configure RG
At the login prompt:
1 2
Login: manager Password: friend Verify confirmation that the upload is complete.
ip set interface ip0 dhcp enabled sntp set server ipaddress [SNTP Server IP Address] sntp set timezone cdt (or edt, mdt, pdt, CST, EST, MST PST--choose appropriate time zone) voip mgcp protocol enable voip mgcp protocol set netinterface ip0 voip mgcp protocol set profile sphere voip mgcp callagent create sphere contact [mgc ip address to which you want this device to check in] voip ep analogue create tel2 type al-fxs-del physical-port tel2 voip ep analogue create tel1 type al-fxs-del physical-port tel1 voip ep analogue enable tel2 voip ep analogue enable tel1 voip ep analogue set tel2 country usa voip ep analogue set tel1 country usa
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AT I R G 6 X X O R I M G 6 X X
Installing
16 17 18 19 20
voip ep analogue set tel2 clip BELL voip ep analogue set tel1 clip BELL voip ep analogue set tel2 codecs g711u,g711a,g729ab,t38 voip ep analogue set tel1 codecs g711u,g711a,g729ab,t38 system config create <filename>.cfg
voip ep analogue set tel1 onhook 1100 voip ep analogue set tel2 onhook 1100 voip ep analogue set tel1 flashhook 600 voip ep analogue set tel2 flashhook 600
--> system config create <filename>.cfg
system restart (to restart the device and apply all the changes)
Logon to the Sphericall Manager to verify the RG device has checked in under the Stations tab. If you have not already assigned extensions or users to this device, follow configuration with Book 1: Plan and Prepare Sphericall Installation and Book 2: Install and Configure Sphericall.
The following commands must be entered in the RG configuration console (or via Telnet) for the analog ports to work propertly :
A new config with a different file name must be created for this command to take effect (cannot have the same config file as before).
3 4 5 6
system config create <file_name.cfg> system config set <file_name.cfg> system config restart
Repeat this sequence for every RG on the system.
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AT I R G 6 X X O R I M G 6 X X
Using
Once this command is entered and set, this sequence of commands will not be required again for the RG.
USING
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The unit is now ready for phone support. Either: Plug in analog phone telephone port. OR Plug in IP phone to LAN port. Refer to IP phone or analog phone guides for phone usage.
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AT I R G 6 X X O R I M G 6 X X
Using
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USB DEVICES
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Eutectics USB Handsets are ideal for integrating with Sphere softphones. With minimal configuration, these handsets are ready for use with the Sphere system.
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EUTECTICS IPP200
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PLANNING
Planning your softphone installation will involve the following: Installation or Sphericall Desktop Installation of phone Installation of drivers for phone, if necessary Reference of the phones user manual for operation information
PREPARING
Be sure to the have following components ready for the installation: Drivers to handle the plug-n-play device. DirectX 8.1 is required on the PC. Any additional executables for the device. In the case of the IPP200, there is an additional set of APIs required for the function of on hook and off hook telephone functions. These files will be included in the final release of Sphericall.
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USB DEVICES
Eutectics IPP520
INSTALLING
1 2 3 4 5 6
Read the instructions with the USB device first. Determine whether you install device or drivers first. Then proceed. Plug in the phoneset (either handset or headset). Update drivers and APIs as required. Sphere software installs will include needed DLLs for some of the Eutechtic phones. Verify Sphericall Desktop software is also installed. Test for functionality. Be sure to read product documents to understand equipment functionality.
USING
Refer to the Sphericall Desktop Manual and Help files for more user information and quick reference guides. Note: You must use the Windows system volume controls to change volume settings.
EUTECTICS IPP520
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PLANNING
Verify system requirements for Eutectics IPP520 and Sphericall. The Sphere system should be installed, configured and tested as fully functional. Refer to the IPP520 User Manual for installation planning, setup, package contents, safety and conditions of use.
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USB DEVICES
Eutectics IPP520
OVERVIEW OF OPERATION
The Eutectics IPP520 phone is a USB phone that can used with the Sphericall Desktop when running as a softphone. Communication between the phone and the Sphericall Desktop is done via the USB interface. A proprietary interface is supplied by Eutectics which allows the Sphericall Desktop to receive messages from the phone as well as send messages to the phone.
INSTALLATION
The drivers for the IPP520 are available at Eutectics web site and are not installed as part of the Sphericall Desktop installation. The drivers are also shipped with the phone. These drivers should be installed when the user first plugs the IPP520 into the USB port on the PC. The dll's that are needed for the interface include the eusbcontrol.dll, eusbevent.dll and eusbi2chook.dll. These files are installed as part of the Sphericall Desktop installation and provide the interface between the Sphericall Desktop and the IPP520. They are installed into the system32 folder.
CONFIGURATION
The IPP520 does not require any special configuration to associate itself with the Sphere system. A softphone line should be created through the Sphericall administrator application and assigned to the user that will be using the phone. When the user starts the Sphericall Desktop, they should verify that the IPP520 is configured as their PC Phone device. This is done through the Configure\Options\PC Phone widow.
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USB DEVICES
Eutectics IPP520
Figure 5.1
At this time the Sphericall Desktop does not get phone specific parameters from the database, such as ring cadences, stutter dial tone, and flash times.
FUNCTIONALITY
Function
Off Hook
Description
When the IPP520 goes off hook the Sphericall Desktop will answer an incoming call or it will display the Dial dialog and play dialtone. If there is already a call in progress it will put the current call on hold before playing dialtone. The user can enter digits through the keypad on the phone or through the dialog. The user can close the dial dialog which will stop dialtone from being played and a call cannot be established until the user goes off hook again. When the user goes on hook the Sphericall Desktop will hang up the current call. The Sphericall Desktop will use the digit map to determine whether it has a valid number before initiating the call. When the user enters digits from the phone the Sphericall Desktop will check the number to see if it can be dialed or if its an invalid number. If it can be dialed the Desktop will initiate a call, if its invalid the Desktop will play fast busy. If the user dials digits over a connected call the digits will be sent over the media stream. The following tones will be played with the IPP520: dial tone, congestion, busy and ringback. Tones not currently supported are outside dial tone, stutter dial tone, call waiting and on-hold reminder.
On Hook Dialing
Progress Tones
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Eutectics IPP520
Function
Hold
Description
The user can initiate a hold from the phone or from the Sphericall Desktop. Pressing the hold button while a call is connected will put the call on hold, pressing the hold button while a call is on hold puts the call into the connected state. The phone provides a mute button which stops audio locally to the phone. The phone does not send a message to the Desktop so it does not know the phone is on mute. If the user selects Mute from the Sphericall Desktop it will mute the call at the Desktop and the phone will not know the call has been muted. A transfer can be initiated by pressing the Transfer key on the phone. When the button is pressed the Sphericall Desktop will display the Transfer dialog if the call is connected or on hold. After the dialog is displayed the Desktop will play dial tone and the user can enter the number using the keypad or using the Desktop. As digits are entered the Desktop will check if the number is valid using the digit map and start the transfer once a valid number has been entered. If an invalid number is entered the transfer will be initiated and congestion will be played. Setting up of a 3 part conference can only be done from the Sphericall Desktop. The phone has a redial button which will send the last number you dialed. This is stored locally. A flash can be initiated using the flash button or by flashing the hook switch for a time less than X seconds. If the user has a call and triggers a flash the Desktop will put the first call on hold and start a new call, another flash will hang up the new call and the first call will be active, another flash and the first call will be reconnected. If the user has more than 1 call the flash will cycle through the calls, creating a new call when a flash is done and the last call is active. If the user then does another flash the first call will be made active. A flash does not initiate a transfer. There are two volume buttons located on the phone just above the microphone. These buttons are active when a call is in the connected state. When these buttons are pressed a message is sent to the Sphericall Desktop which then changes the volume settings for the audio device. The IPP520 has a speaker and external microphone which allows the user to operate the phone as a speaker phone.
Mute
Transfer
Volume
The phone has memory keys which allows for storage of frequently used numbers. Since these numbers are stored in the phone the Sphericall Desktop has no knowledge of these numbers. A headset can be connected to the IPP520. Star codes are not supported with the IPP520. The Sphericall Desktop must be running for the IPP520 to function so the Desktop is used rather than the star codes.
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USB DEVICES
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PLANNING
Verify system requirements for Plantronics CS50-USB and Sphericall. The Sphere system should be installed, configured and tested as fully functional. Refer to the CS50 User Manual for installation planning, setup, package contents, safety and conditions of use. Verify necessary firmware for base and ear piece.
OVERVIEW OF OPERATION
The CS50-USB offers wireless, hands-free headset convenience and long range workspace mobility.
INSTALLATION
Consult the manual shipped with the CS50-USB unit.
1 2 3 4 5 6 7 8 9
Install the hardware. Install the software. Charge your headset battery. Choose your headset wearing style. Carry out the initial setup recommended by the manufacterer. Open the Sphericall Desktop application. Go to Configure, Options. Select the Plantronics tab. Make your usability choices.
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USB DEVICES
Figure 5.2
Operation The Sphericall Desktop interface opens the Place New Call window when the Plantronics CS50 headset is taken out of the charging cradle.The user is able to get dialtone, answer calls, or drop calls all with 1 click. The user can click the button once when there is an incoming call and the call would be answered. Also, when the user takes the headset out of the cradle, the Sphericall Desktop obtains a radio link established message and handles it accordingly, it provides dial tone if there is no call, or answers an active incoming call.
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USB DEVICES
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Theory of Operation The Windows Messenger application has gained such a popular reputation for instant communication that some organizations are incorporating other protocols within the messenger interface. Sphere Communications supports Microsoft Windows Messenger v5.1.0701 on Windows 2000 or Windows XP application integration to its communication platform. In addition to the convenience of full-text messaging offered by Windows Messenger, users can add the operability of voice communications from the same user interface. Windows 2000 Windows 2000 clients may be required to uninstall versions of MSN Messenger in order for the Windows Messenger to operate. Only voice calls are supported; not video. Windows XP Voice & Video are supported. Windows Messenger supports SIP (RFC3261 and 3428) and the SIMPLE extensions for presence and instant messaging. The SIP protocol is used to communicate with Sphericall for establising and maintaining voice and video connections with other Sphericall endpoints. The Windows Messenger user must configure a SIP communications service account to: Identify the Sphericall Manager that will act as a SIP server for this Messenger client. Identify the Sphericall Manager name and/or IP address. Identify the appropriate connection protocol (UDP). Once configured properly, Windows Messenger can be used as a softphone interface as well as a text messaging interface. No other application or program is needed.
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PLANNING
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The following items are needed in planning for a system using Windows Messenger clients for the softphone:
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Preparing
As always, refer to the Sphere System Requirements for version compatibility and interoperability notes. Messenger clients are limited to handling a single voice or video call at a time. An integrated environment with Messenger and Sphericall Desktops allows growth for some users past the limit of a single call. Messenger clients can be configured with a Class of Service (CoS) profile. Permissions can also be used with Allow and Disallow on the Sphericall Administrator. All Messenger users must be assigned a primary numeric extension number. This allows a number to which hardware phone users may call. Forwarding is not configurable at the Messenger client level, rather, it can only be configured by the Sphere system administrator on Sphericall Administrator. For a Sphericall Desktop user to be able to text message a Windows Messenger user, the administrator must set up privileges for the Desktop user to monitor the line of the Instant Messenger recipient. Video is not enabled by default for SIP Windows Messenger clients. You must adjust an initialization setting to enable this. Please refer to the Sphere System Requirements for information regarding SIP and video.
PREPARING
REQUIRED:
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Computer/Processor (as specified by Microsoft, see their site for the most up-todate requirements): Computer with 300 megahertz or higher processor clock speed recommended. 233 MHz minimum required (single or dual processor system). Intel Pentium/Celeron family, or AMD K6/Athlon/Duron family, or compatible processor recommended. Additional requirements for video and/or application sharing Windows Messenger Version 5.1.0701 on Windows XP or download for Windows 2000. Speaker and microphone device for voice communciations OR USB headset or handset.
INSTALLING
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Note: The installation procedure for Windows Messenger should be followed by the
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Installing
Logon to Sphericall Manager. Open the Sphericall Administrator application. Click on the General tab. Highlight the top tree level System [SERVER NAME]. Double-click to open properties or right-click and select View Properties. Enter the SIP Domain name into open field: Session Initiation Protocol. Create extensions and fill in the user information (First Name, Last Name, etc.).
Choose Tools/Options/Accounts.
Figure 6.1
3 4
Enter the user information in the SIP Communications Service Account field. Click the Advanced tab in that field.
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Installing
Figure 6.2
5 6 7 8 9 10
Select Configure Settings. Enter the name or IP address of the Sphericall Manager to which this Windows Messenger will check-in. Click on UDP in the Connect Using area. Click OK. Click OK again to exit Options. Sign-in with Windows Messenger connecting to the Sphericall Manager.
Set Windows Messenger security policy to Low Security. The following registry key must be added to each PC running Messenger
HKLM\Software\Policies\Microsoft\Messenger\Client\{83D4679F-B6D7-11D2-BF3600C04FB90A03}\_Default\EnableSIPHighSecurityMode DWORD= 0 -- Low Security 1 -- High Security 2(Default, same as not set) --- Medium security" For ease of deployment, this registry setting can be wrapped in an executable registry file. This file is available on the Sphericall software DVD under: \Client\Messenger\MessengerPolicy.reg. The system administrator may configure his network so that this file is executed when users log into the domain.
3
You will need to return to the Sphericall Manager to configure this Messenger Clients primary Extension number at this time.
Return to the Sphericall Manager. Open the extension number which you have assigned this user.
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Installing
Add the SIP address, which has now checked-in on the Sphericall Manager, to the Numbers area of the Properties for this Station.
This step must also be completed for regular phones without messenger to make calls to this Windows Messenger client.
4
Verify that the Sphere system assigns this extension as the primary extension.
Click on the Settings tab. Click Add to add a new Initialization Setting for this Messenger client. Select SIP > Video. Change the Value from Disabled to ENABLED. Click OK.
This is the customer message users will see when users are added to their Windows client. This dialog only appears the first time the Messenger line checks into a Sphericall Manager. If the Messenger line is deleted from Sphericall and then re-added through another check-in, this dialog box may not appear.
Note: You may need to log off and log back on in order to get this message.
1
Click OK.
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Testing
TESTING
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After installing the Windows Messenger client, the Sphere system administrator should test and verify that the application is consistent in the following areas: Always verify from the Sphere System Requirements that the correct version of Microsoft Windows Messenger is being used the the Sphericall software. Send and receive an audio call to/from another extension in the Sphere system. Set up a conference call where the Windows Messenger client is a member. Conduct blind and attended transfers from the Windows Messenger client to another station. Have the Windows Messenger client send and receive video. Have the Windows Messenger client send and receive an Instant Message.
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Note: When calling another station from Windows Messenger, the caller does not
hear ringback.
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Figure 6.4
Figure 6.5
Notification of change
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MUSIC O N H OLD
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MUSIC-ON-HOLD
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This chapter is divided into three sections:
Section I - Hardware-Based Music-on-Hold Section II - Music-on-Hold Installation Test Section III - Music-on-Hold and Zones
MUSIC-ON-HOLD REQUIREMENTS
Hardware-based Music-on-Hold sources are supported. However, both hardware and media server MOH will not be supported at the same time. You must configure a station as a MOH input port via the Sphericall Administrator application. Your organization must secure a music source with a standard audio output jack (such as a CD/DVD player).
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MUSIC ON HOLD
Music-on-Hold
A line within a zone must be designated as the MOH source for the zone. All zones with independent MOH must have individual physical equipment for their audio source. Each zones physical audio source must correspond to a distinct station port on a PhoneHub or BranchHub. All re-broadcast rights must be purchased according to FCC statutes. Your organization must secure a third-party interface device that will connect the standard audio output jack at the music source to a station (i.e. PhoneHub port). An example of this type of interface device is the Bogen WMT-1A Telephone Line Matching Transformer. MOH Requirements Multicast (wherever available) Unicast (as necessary) MOH must be assigned to a Sphere system zone. Be sure to refer to the Sphere System Requirements for UDP port information for system-wide MOH multicast address.
Figure 7.1 EXAMPLE: Bogen WMT-1A Telephone Line Matching Transformer
Audio Source
Refer to the following table for a short list of distributors that manage an inventory of these requisite, third-party interface devices. Contact your organizations Sphere Certified Channel Partner or Sphere Communications sales representative, if you require further assistance in locating a local supplier.
Table 7.1 Third-Party Interface Device Distributors
Distributor
Jenne Distributors ALLTELL Supply Sprint North Supply
Telephone Number
+1 (800) 422-6191 +1 (800) 725-5835
Internet Address
www.jenne.com www.alltell.com www.sprintnorthsupply.com
RE-BROADCASTING RIGHTS
FCC statutes define a corporation or other organization as a place where a substantial number of persons outside the normal circle of a family and its social acquaintances is gathered (17 U.S.C. SECTION 101). Corporations and
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MUSIC ON HOLD
Music-on-Hold
organizations are considered public places that must adhere to copyright guidelines regarding the performance of recorded audio and video materials. Whether the performance is the use of a Music-on-Hold device (which constitutes a public performance by virtue of its being a transmission to the public), the playing of a CD/DVD, MP3 or tape, or the tuning of a radio to a particular station, permission must first be obtained in order for the use of that performance to be considered lawful. Once an organization purchases a Music-on-Hold source, they have full rights to rebroadcast music via that device and no other royalties are due the vendor. Radio broadcasts throughout a Sphere system are permissible if licenses have been purchased for each incoming telephone (i.e. trunk) line.
station ports. Sphericalls hardware-based MOH feature can support both a single music source for an entire system, as well as individual music sources for multiple zones within the system. Prior to Sphericall v3.4.4.4, all MOH sources were based in the COHubs Zone, meaning the inbound call determined what MOH music was heard. With v3.4.4.4, the MOH is determined by the person placing the call on hold from within the system; MOH for that zone will reflect that stations zone (i.e. MOH could include time of day, weather, etc. reflecting that called stations zone rather than the COHub from which the call entered the system). All zones must have independent physical resources for MOH. If a system requires three unique zones of MOH and a default zone with Music-on-Hold for fallback, there must be four physical MOH sources to meet this need as well as a separate station port for each MOH source. If a system needs only one MOH source, only one physical source is needed. All MOH music sources may be located on one PhoneHub or BranchHub, however each source must be assigned to a separate zone. One default zone may serve all MOH needs. If individual MOH needs are present, then the system must be designed for MOH by Zones. In order to design your system for optimal MOH service to meet all zone needs, you will need to consider all physical requirements. If a specific MOH source is not configured for a zone, the MOH audio source of the default zone will be used. Systems with a large number of zones may only require a few specific MOH sources. Rather than force MOH to be configured for each zone, the fallback to the default zone will ease configuration and minimize cost. Use the following instructions for configuration of one MOH per zone, as needed for the system you are configuring.
Click the General tab. Select Music On-Hold Sources from the tree.
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MUSIC ON HOLD
Music-on-Hold
Figure 7.2
The Music On-Hold Sources branch will contain either a list of hardware station lines, established from a pre v4.1 version of Sphericall or a Sphericall Media Server(s) entry. By default, a new media server is initialized to be a source for MOH . Configuration or loading may necessitate barring a media server from sourcing MOH. Media servers can be barred from sourcing MOH. Assuming that this is a new installation and there are no previously-established Music On-Hold Sources:
3 4 5 6 7
DELETE the MOH Media Server listed here (you will not need it if you are using hardware based MOH). Click OK to exit this window. Highlight Music On-Hold Sources. Right-click to Add. Click Add.
Adding a new MOH Source when the list is blank will display a dialog box asking the user to choose either media server or hardware-based MOH.
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MUSIC ON HOLD
Music-on-Hold
Figure 7.3
8 9
Select the zone with which to associate this station from the Zone list. Expand the MG(s) listed in the display area to view all available stations within your organizations Sphere system. Highlight the station you wish to configure as the MOH line. Click OK.
The selected station appears in the Music On-Hold Line display area.
14 15
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Music-on-Hold
Figure 7.4
16 17 18 19
Type the name of this Music-on-Hold line in the Line Name field. Select the In Service check box if you wish to classify this station as active within your organizations Sphere system. Clear the Log Calls check box. Select the Drop Loop Current check box.
Drop loop current is primarily used for paging and voice mail lines. For every station on a PhoneHub or BranchHub that is connected to a paging system, that stations properties must be configured in this manner.
20
Verify that the Music-on-Hold extension, moh1, is listed in the Numbers area. Click Add Extension. Expand the folders listed in the file tree to view all available extensions within your organizations Sphere system. OR Click New Extension to create a new extension for this purpose. Note: Because the moh1 group extension cannot be made primary for this station,
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MUSIC ON HOLD
Music-on-Hold
will serve as a primary extension for the Music-on-Hold station and associate it with this station.
4 5
Highlight the extension you wish to assign to this station. Click OK.
The newly-assigned extension appears in the Numbers area in the Properties for Station window.
6 7
Highlight any assigned extension number (the one you just created) besides the moh1 extension. Click Make Primary for this new extension number (do not make MOH numbers primary).
Primary extensions appear in bold type in the Numbers area as well as in the Stations tab.
CAUTION!
8 9
GROUP EXTENSIONS AND VOICE MAIL EXTENSIONS ARE NEVER TO BE MADE PRIMARY EXTENSIONS WITHIN A SPHERE SYSTEM.
The configured Music-on-Hold station appears in the Music Hold Line display area.
Place an external call from your organizations Sphere system to an address located within the Sphere system.
Table 7.2
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MUSIC ON HOLD
Music-on-Hold
One Existing Hardware-based Musicon-Hold source and more than one zone
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Music-on-Hold
Hardware-based MOH: more than one zone with more than one Music-on-Hold and with no Music-on-Hold source located in the default zone
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MUSIC ON HOLD
Music-on-Hold
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PAGING
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PAGING LINES
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OVERVIEW
Creating a paging line provides users with an opportunity to dial an extension and page their party over your organizations intercom or public address system. These paging lines are used to integrate separately-purchased, third-party systems with a Sphere system. The paging line is configured, similar to any other station, via the Sphericall Administrator application and is connected to a station interface on either a PhoneHub or BranchHub.
RECOMMENDED PRODUCTS
All paging station adapters must have the capability to disconnect on dialtone. A product that has been Sphere-tested is the V-9970 Station-Level Page Adapter from Valcom, Incorporated: +1(540)427-3900 or http://www.valcom.com The V-9970 can be enabled for this disconnect capability by setting its dipswitch #1 to the On position. This prevents inadvertent dialtone broadcasts over your organizations paging system. Common Paging System Features
Table 8.1 Paging System Features
Feature
Talkback
Description
If your organizations Sphere system has been integrated with the appropriate paging system hardware, you can configure paging for two-way voice communication over that system. Note: This configuration is commonly used for security and entry-management purposes. Many paging systems offer background music inputs that allow the paging system to play music over the PA when not in use. Some paging systems can announce incoming calls to any address throughout the entire system via ring tones played over the PA. Note: This configuration often requires additional system hardware. Consult the documentation included with the third-party paging system for more information on this feature functionality.
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PAGING
Paging Lines
Feature
All Call
Description
Additional extensions can be connected to the paging system to enable the simultaneous paging of all zones within your organizations Sphere system. Note: This configuration is commonly used with multi-zone paging systems utilizing multiple paging extensions.
organization intends to integrate such a paging system with their Sphere system, you must secure a station-level paging adapter to provide the necessary call disconnect functionality. Call disconnect functionality can be provided with either an open-loop detector (i.e. drop loop current), a time-out release (per the time frame specified for the paging device), or an audio-sense release (i.e. the paging system disconnects the call when it detects no sound on the line).
Click the Number Plan tab. Click Add Extension Number. Type the paging extension number in the Number field. Select Single Line from the Hunt Order list. Type the name of this paging extension in the Last Name field. Select the Show in Phonebook check box if this extension is to appear in your organizations Sphere system extension list.
Select the Paging Device icon type to associate with this extension.
By selecting the Paging Device icon type, the extension/station combination will play a different dial tone than a normal station.
8 9
Click Apply. Click New. and Repeat the previous steps to add another paging extension to your organizations Sphere numbering plan. Click OK when you are finished adding paging extensions to your organizations numbering plan.
10
Sphere numbering plan before you can add and configure a paging line (i.e. station). From the Sphericall Administrator window:
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PAGING
Paging Lines
1 2 3
Click the General tab. Select Paging Lines from the list. Click Add.
Expand the MG(s) listed in the display area to view all available stations within your organizations Sphere system. Highlight the station or BranchHub line you wish to designate as the paging line. Click OK.
Figure 8.1
10 11 12 13 14
Type the name of this paging line in the Line Name field. Select the zone with which to associate this station from the Zone list. Select None from the Pickup Group list. Select the telephony area with which to associate this station from the Telephony Area list. Select the emergency group with which to associate this station from the Emergency Group list.
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PAGING
Paging Lines
15
Select the paging line profile from the Default Profile list. Note: Sphere Communications strongly suggests that you create a separate paging
line profile with Max Calls set to 1 (to prevent more than one call to access the PA system at one time) and with call waiting disabled (to negate the ability to interrupt an in-progress page).
16 17 18
Select the In Service check box if you wish to classify this station as active within your organizations Sphere system. Clear the Log Calls check box. Select the Drop Loop Current check box.
Drop loop current is primarily used for paging and voice mail lines. For every station on a PhoneHub or BranchHub that is connected to a paging system, that stations properties must be configured in this manner.
19
Verify that the paging extension number appears in the display area. Highlight the paging extension. Click Make Primary. Click Apply. Click OK.
Connect the tip-and-ring of the 6-position modular jack to the tip-and-ring inputs for the adapter on the punchdown block (or breakout box/patch panel) serving the PhoneHub (or BranchHub). Note: The RJ-11 wiring arrangement is:
Refer to the documentation included with the third-party paging system for information and instructions regarding the connection and configuration of speakers and other associated hardware.
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PAGING
Paging Lines
Punchdown Block
Station-Level Paging Adapter Single-Zone Paging System
Zone 1
Sphere MG
Zone 2
Zone 3
Sphere MG
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Paging Lines
Zone 1
Zone 2
Sphere MG
Zone 3
INSTALLATION TEST
To verify successful paging system installation and configuration
1 2 3
Confirm the appropriate hardware installation for the third-party paging system. Dial the paging extension configured for your organizations Sphere system. Announce the page for the appropriate paging zone.
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SMDI AND SPHERICALL VOICE MAIL
SMDI and Sphericall Voice Mail are independent of one another. Each voice mail system can be accessed independently by using unique addresses. A Sphere system can use either its native Sphericall Voice Mail, SMDI integration of voice messaging with another application, or both. A station can receive message waiting indications from either an SMDI or Sphericall Voice Mail system. Stations that are using both Sphericall and SMDI voice mail systems may get their MWI state out of synchronization. SMDI may clear the MWI regardless of any unread Sphericall messages. The preferred configuration is to limit a station to single voice mail system.
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SMDI OVERVIEW
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A Sphericall systems integration with an SMDI platform utilizes the Simplified Message Desk Interface and conforms to an industry-standard method of voice messaging platform integration (via the RS-232 serial connection found on any personal computer). Because the information about the telephone calls travels along a different channel than the calls themselves, the Sphericall/SMDI integration is considered an out-ofband integration. This differs from an in-band or DTMF integration where telephone call information is passed via the same channel as the call. Any service utilizing SMDI integration can be used with the Sphere system, for example, all of the following can integrate with Sphere via SMDI: Voice Messaging Call Center Fax Center
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Sphericall also supports the transmission(s) of the following messages within an integration:
Table 9.1 Sphericall/Voice Mail Integration Message Transmission
Messages Sent to Voice Mail System
Direct call Forward all calls Forward on no answer Forward on busy Forward with unknown reason Invalid message waiting request Blocked message waiting request
MANUALS
Detailed manuals for Sphericall and other third-party voice application products may be necessary to determine full integration capabilities as well as instructions for installation. All Sphere System Requirements as well as the SMDI requirements are also contained in the Sphere System Requirements manual.
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Voice messaging-related portions of the Sphere system adhere to the Telcordia (formerly Bellcore) standard. Refer to Telcordias web site at http://www.telcordia.com for more information. Sphericall integrates with third-party SMDI capable voice mail applications.
Table 9.2
Requirement Sphere SMDI Hardware Requirements
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Requirement
Sphericall Manager
Description
The Sphericall integration relies heavily upon the communication between the two server types, Sphericall Manager and the other third-party voice server. The Sphericall Manager houses the Sphericall software, the engine that ultimately drives the integration of the systems. Note: A single Sphere system can support one voice messaging platform per SMDI link. Each Sphericall Manager must have its own set of voice messaging ports and its own RS232 serial port link. As such, a separate Sphericall Manager will be used for each voice messaging platform connection. Each MG must have the appropriate number of available ports. The Sphericall integration passes information between servers (Sphericall Manager and other voice server), and this information is ultimately passed to the endpoints (i.e. telephones attached to MGs) on a Sphere system.
Data Bits Parity Stop Bits Hardware Flow Control Link Operations
Digit Length
Cabling Requirements
Sphere PhoneHub and BranchHub Cables Refer to the PhoneHub Installation Manual or the BranchHub Installation Manual for complete MG cabling requirements. 1 (one) RJ-11 two-wire or RJ-14 four-wire telephone cord is required: Per station port Per 50-pin connector depending upon the platform connection requirement. 1 (one) available RS-232 compliant serial port is required per SMDI integration. The serial port may be either 9- or 15-pin. However, you must have the appropriate cable adapters if you plan to mix the two types. The serial ports on the Sphericall Manager and the third-party voice server are both configured as DTE (Data Terminal Equipment) devices. The RS-232 (i.e. serial-to-serial) cable, otherwise known as a Null Modem or cross-over cable, is utilized to connect the Sphericall Manager to the third-party voice server. The cable should be of sufficient length to reach between the Sphericall Manager and the third-party voice server but should not exceed a 50-foot distance limitation for the RS-232 connection.
RS-232 Cable
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Platform recommendations/considerations
Description
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SMDI Operation
SMDI OPERATION
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OVERVIEW
A Sphere system is integrated with third-party voice messaging systems in the following manner:
Figure 9.1 SMDI-to-Sphericall Connections
Switch BranchHub
PSTN
COHub
SphericallManager
PSTN
Switch
PhoneHub
RJ-11 or 50-pin
Configuration
NT Control Panel, startup and shutdown to Telephony Server etc
In order to process an incoming call, a voice messaging system needs certain pieces of information about the call: The voice messaging system needs to know the called party ID. This is the extension number to which the call was forwarded by the voice mail system. The voice messaging system can also use the calling party ID. This is the extension or telephone number of the party placing the call. The voice messaging system needs a way to communicate information to the telephone system (i.e. the correct status of the message waiting indicators that correspond to the subscribers mailboxes).
SYSTEM INTEGRATION
The Sphericall/SMDI integration is considered an out-of-band integration. In such an integration, information is sent between the telephone system and the voice messaging system on a dedicated data link, generally an RS-232 format link. When the voice messaging system answers a forwarded call from the telephone system, the telephone system has also sent the relevant integration information to the voice messaging system across the data link. When the voice messaging system wants to notify the telephone system of a change in MWI status for a specific user, it uses the same data link.
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SMDI Operation
CALL PROCESSING
THE COMPLETE INTEGRATION
Because the Sphericall/SMDI integration is an out-of-band integration, some form of synchronization must occur to link the out-of-band message packet to the appropriate telephone call. This synchronization occurs as a result of the interaction of two separate entities, the Message Storage Retrieval Identifier and the Message Storage Retrieval Line ID. The MSRID is the unique identifier for the voice mail platform within a Sphere system. The MSRLineID is the name given to an individual analog station port connecting the Sphericall Manager and the voice mail server.
Note: You must obtain all of the port numbers from the voice mail server prior to
configuring the Sphericall Softswitch. Examples and Functionality Refer to the following examples to determine the relationship of Sphericall addresses to MSRLineIDs.
Table 9.4 Address Correlation to MSRLineIDs
VM Address
6000 6001 6002 6003 6004 6005 6006 6007 6008 6009
MSRLineID
N/A (Group Number) N/A (Group Number) 0001 0002 0003 0004 0005 0006 0007 0008
VM Group Number
6000 * 6000 6000 6000 6000 6000 6000 6000 6000
AA Group Number
* 6001 6001 6001 6001 6001 6001 6001 6001 6001
If an analog station port identified as MSRLineID 0001 on a Sphericall Manager was connected to a port that the voice mail server identifies as MSRLineID 0002, integrated calls will not travel between the two servers. Calls will traverse the RS-232 serial cable connection, and the voice mail server will be notified that it is to pick up the call at the port identified as 0001. However, the voice mail server will be unable to retrieve the correct call because the Sphericall Softswitch sent the call to the station port identified as 0002. These matching requirements hold true for the MSRID as well. All IDs must match or the voice mail server will be unable to process the telephone calls.
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SMDI Operation
SPHERICALL
A Sphere system presents two general types of calls to voice mail. The first call type is an auto attendant call. Auto attendant calls are directed to voice mail via telephone system programming: Voice mail voice ports are programmed into a hunt group within Sphericall, and specific trunks are programmed to ring that hunt group. Sphericall then finds an available voice mail port and presents the call to the voice messaging system. The second type of call presented to voice mail is a call forwarded from a station. Individual stations within Sphericall can be programmed to forward to the voice mail hunt group if the station does not answer within a specific number of rings (a Ring No Answer Forward). Regardless of call type, the Sphericall Softswitch delivers the call to voice mail along with the required integration information. The voice mail system uses this integration information to determine how to handle the call.
Description
An internal telephone extension calls directly into voice mail An internal telephone extension calls another internal telephone extension and is forwarded to voice mail An outside call is sent directly to voice mail An outside call is directed to an internal station and is forwarded to voice mail The telephone system operator initiates a call to voice mail
A voice mail systems integrated messaging function will process each of these call types differently: In the case of a forwarded station or forwarded trunk call, voice mail routes the call to the Subscriber mailbox associated with the called party ID extension number. In the case of a direct trunk call, voice mail acts as an auto attendant and routes the caller to the appropriate mailbox. If callers dial an extension number via one of the call processor menu options, voice mail attempts to transfer the call to the appropriate extension. If that extension returns Busy or RNA CP tones, a voice mail system, acting as a voice messaging system, prompts the caller to record a message for the subscriber. In the case of a direct station call, voice mail prompts the caller to leave a voice or fax message for the subscriber or, if the caller is a subscriber, prompts the caller to enter the security code for the mailbox.
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If the caller leaves a message, voice mail routes the message to the appropriate mailbox and uses the telephone system to light the MWI on the telephone set associated with the appropriate subscriber. If the caller accesses the mailbox, the voice mail system provides a prompt for mailbox administration and maintenance. Subscribers can listen to, delete, and send messages as well as record personal greetings and name recordings for their mailbox.
THE SUBSCRIBER
When voice mail subscribers notice that their MWI is lit, they may pick up the telephone and dial into voice mail to retrieve messages. When the voice mail system recognizes this call type (a direct station call), it uses the calling party ID to identify the mailbox associated with the extension placing the call and prompts subscribers to enter their security code. After they enter their security code, voice mail places them into their mailbox. Once in their mailbox, subscribers are presented with a menu that, among other choices, allows them to process any new messages. If subscribers choose to delete the messages after listening to them, voice mail again uses the telephone system integration to extinguish the MWI on the appropriate telephone sets.
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PLANNING FOR VOICE PORTS
Utilizing the Erlang model, system integrators must estimate the number of IVR ports an organization needs in order to connect the Sphericall Manager(s) to a third-party voice messaging system. This analysis and subsequent planning prevents unacceptable call blocking based on a systems lack of IVR ports.
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Indicator of the amount of call volume depending on an organizations work flows. An organization with flexible hours would have a smaller multiplier than the business with set hours. An organization with a heavy call volume would have a lower multiplier than an organization with light traffic. Typical values are 1.5 to 2.5. A fraction representing the calls that cannot be completed because all lines are busy. If blocking = 0.1, 10% of calls are blocked. Depending upon the application, reasonable figures for blocking are between 0.01 and 0.05. Callers experiencing blocking usually hear a busy signal. A measurement of telephone traffic. An Erlang is equal to one full hour of use (e.g. conversation), or 60 * 60 = 3600 seconds of phone conversation. CCS is converted into Erlangs by multiplying by 100 then dividing by 3600 (i.e. dividing by 36). Numerically, traffic on the trunk group (when measured in Erlangs) is equal to the average number of trunks in use during the hour in question. Note: In the US, the unit commonly used for traffic measurement is CCS (centi-call second). The following conversion factor can be applied between the two units: 1 Erlang = 36 CCS 1 CCS = 0.0278 Erlangs Common term referring to the physical communication line between two switching systems such as a local PBX and the telephone companys CO switch. The number of trunks is a figure represented by the number of lines in a trunk group. An IVR systems electrical interface through which a PBX sends information to a voice mail system. The system design needs to account for the potential volume requirements of port interfaces between the PBX and the voice mail server. The number of IVR ports is a figure represented by the number of station ports necessary on the system.
Blocking
Erlang
Trunks
IVR Port
Call loggers
Many PBXs have call loggers connected to them. These record and analyze details of calls made using the PBX and can provide traffic figures for particular routes. To ensure that the figures used are busy hour figures: Verify the busy hour figures as provided by the call logger itself Print hourly traffic figures over a period of time then use the highest hourly figure over that period Convert call figures from minutes to Erlangs To convert from hourly call minutes to Erlangs, divide the figure by 60. Remember that call loggers are aware only of calls that were successfully completed (i.e. call loggers provide carried figures rather than offered figures, which your calculations require); this is only an issue if the blocking on the trunk groups is already a problem
2
PBX statistics
Some PBXs (i.e. Nortels Meridian PBX) have software modules provided that provide the required traffic figures.
Note: The same considerations as for call logging figures apply here.
3
Telephone bills
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You can use telephone bills, sometimes, to discover the traffic carried by a trunk group.
4
For a fee, your existing carrier may be prepared to complete a study of your organizations call traffic and may be able to use historical data.
5
Estimating
If you do not have any of these luxuries, you must make a reasonable estimate based on what you know about the way the organization uses its telephones; you may also be able to use information from other similar sites. A table is provided concerning the estimation of Erlangs for trunk and IVR usage.
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Table 9.7
of ports for the integration. For example, if your organizations voice message system requires 8 ports (or a multiple of 8 ports), Sphere system setup is simplified and troubleshooting, minimized, if you designate for service an entire PhoneHub or a portion of a PhoneHub.
INTEGRATION NOTES
Utilize the information in the following sections as appropriate during configuration of your organizations Sphericall and voice messaging integration.
Frequency
400 Hz 350 Hz + 440 Hz 480 Hz + 620 Hz
Cadence
Continuous Continuous On = 0.5 s Off = 0.5 s On = 2.0 s Off = 4.0 s 0.25 s, 0.4s, 0.4 s NA
Amplitude
-27 dBm -24 dBm -24 dBm
Ringback Tone
440 Hz + 480 Hz
-24 dBm
-24 dBm NA
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Frequency
NA
Cadence
Amplitude
-24 dBm
Amplitude levels: Measured on the PhoneHub with HP 3558 Transmission & Noise Measuring Set relative to 600 Ohms. Acceptable Flashtime (range): 350-1100 ms
IVR Line Caller ID = +1 8475551234 Caller A Caller ID = +1 8475551234 Caller B Caller ID = 5011 Caller ID is that of Caller B until Caller B hangs up to connect the call. Then, caller ID information passed is that of Caller A. Initial Caller ID = 5011 Caller C
Configuration\Hardware\Switch Protocol tab. Set the value for the Transfer Init field to FP*96. The ivrline Line Setting You can also configure signalling over the serial port connection as a terminal settingivrline with a value of yesfor each of the IVR lines configured within the Sphere system. Beginning with Sphericall v3.2, the Sphericall Softswitch uses the correct caller ID for calls regardless of whether FP *96 is passed between the voice messaging platform
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and the Sphere system. This terminal setting notifies the MGC that a particular line is used for voice messaging functionality within the integration. Now, when an IVR line places an attended transfer call to a recipient party, the caller ID that is displayed is the caller ID of the original calling party, not the caller ID of the transferring station (as depicted in the following figure).
Figure 9.4 The ivrline Line Setting and Caller ID Presentation
IVR Line Caller ID = +1 8475551234 Caller A Caller ID = +1 8475551234 Caller B Caller ID = 5011 Caller ID is that of Caller A. W hen Caller B hangs up to connect the call, the caller ID inform ation passed is still that of Caller A. ivrline term inal setting here Initial Caller ID = +1 8475551234 Caller C
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SMDI INSTALLATION
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10
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Have you installed, configured, and ensured network connectivity to all of the appropriate Sphere MGs and IP phones for your organizations Sphere system? Have you configured the general PBX properties and functionality for your organizations Sphere system? Have you defined the numbering plan for your organizations Sphere system? Have you defined all of the global and local system settings for the Sphericall Manager? Have you installed, configured, and integrated the Sphericall Desktop on workstations throughout your organizations Sphere system?
Prepare a Sphere system for SMDI integration with the voice mail platform Configure the Sphere system to support voice mail functionality
a. Configure Sphericall extensions as voice mail extensions b. Configure voice mail and auto attendant hunt groups
3 4 5
Enable a message store that will serve as the intermediary between a Sphere system and voice mail Test and troubleshoot the SMDI integration Interpret SMDI messages passed within an integration
NOTES
Much of the information contained in this chapter is intended to serve as a quick reference point for integrating a Sphere system with a voice messaging platform. The more detailed, site- and system-specific information is contained in specific product manuals, for example refer to the your third-party voice message server documentation AND Sphericall manual for specifics SMDI integration.
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SMDI INSTALLATION
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Before installing and configuring the voice mail server as your organizations voice messaging solution, you need to prepare your organizations Sphere system for integration with the new voice mail server.
Click Start\Settings\Control Panel. Double-click Ports. Highlight COM<X> where X is the number of the appropriate serial port to be used for platform integration. Click Settings.
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6 7 8 9 10 11
Select 9600 from the Bits per second drop-down list box. Select 7 from the Data bits drop-down list box. Select Even from the Parity drop-down list box. Select 1 from the Stop bits drop-down list box. Select Hardware from the Flow control drop-down list box. Click Advanced.
12
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Click the Primary Sphericall Manager or Secondary Sphericall Manager radio button to declare if this Sphericall Manager is to be considered a Primary or Secondary Sphericall Manager.
There can be only one Primary Sphericall Manager in the Sphere system.
Note: If you selected the Secondary Sphericall Manager radio button, you will be
asked to enter the Primary Sphericall Manager for your organizations Sphere system. Otherwise, the commissioning process is the same.
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4 5
Select the appropriate check box if this Sphericall Manager will be connected to a voice mail system through a serial port (i.e. an SMDI link). Click Next.
Click Finish.
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When integrating voice messaging and auto attendant systems with your organizations Sphere system, you should establish a numbering plan that incorporates two lead numbers (one for the voice mail and one for the auto attendant) as well as the extensions used with the appropriate voice mail ports:
Voice Mail Auto Attendant Hunt Extensions
Extension the Sphere system uses for voice mail (example: 4000 or 500). Extension the Sphere system uses for auto attendant (example: 4001 or 501). Primary extensions the Sphere system uses to enable the voice messaging ports (example: 4005, 4006, 4007, etc. or 504, 505, 506, etc.).
In accordance with your system integration design, you should create and configure the requisite extensions and stations only after you enable the SMDI service via the Sphericall Manager Configuration utility. Once you enable the SMDI service, you should then create and configure all of the phantom extensions that will serve as station ports for voice mail lines (i.e. 6000, 6001, 6002, 6003, etc.). The number of extensions required for your organizations Sphericall/SMDI integration depends upon the number of ports to be connected to the voice mail server.
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Figure 0.1
A voice mail extension can never be the primary extension on any station. If only one extension exists on a station, that extension, by default, is made the primary extension. Phantom extensions are created to be the primary extensions associated with the stations.
4 5
Select Single Line from the Hunt Order list. Type the name of the extension in the Last Name field.
Clear the Show in Phonebook check box to prevent this phantom extension from appearing in your organizations phonebook. Select the Unknown icon type. Click Apply. Click OK.
Click the Stations tab. Expand the MG(s) listed in the display area to view all available stations within your organizations Sphere system. Highlight the station you wish to configure. Click Station Properties.
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5 6 7 8 9 10
Type the name of this station in the Line Name field. Select the zone with which to associate this station from the Zone list. Select None from the Pickup Group list. Select the telephony area with which to associate this station from the Telephony Area list. Select the emergency group with which to associate this station from the Emergency Group list. Select VM Port from the Default Profile list.
The VM Port default profile assigns the following settings for the line:
Max Calls = 1 Outside Calls Allowed No Call Waiting
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Select the In Service check box if you wish to classify this station as active within your organizations Sphere system. Clear the Log Calls check box. Select the Drop Loop Current check box.
Drop loop current is primarily used for paging and voice mail lines. For every station on a PhoneHub or BranchHub that is connected to a port on the voice mail platform, that stations properties must be configured in this manner.
14
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Click Add Extension. Expand the folders listed in the file tree to view all available extensions within your organizations Sphere system. Highlight the appropriate phantom extension you wish to assign to this station. Click OK. Click Add Extension. Expand the folders listed in the file tree to view all available extensions within your organizations Sphere system. Highlight the appropriate voice mail extension to be assigned to this station. Click OK. Click Add Extension. Expand the folders listed in the file tree to view all available extensions within your organizations Sphere system. Highlight the appropriate auto attendant extension to be assigned to this station. Click OK.
The three voice mail, auto attendant, and phantom voice mail extensions are now associated with this station and appear in the Numbers area.
13 14
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2 3 4 5 6 7 8
Click Add. Click the highlighted line under Name. Select Set As Voice Mail from the list. Click the adjacent line under Value. Select true from the list. Click Apply. Click OK.
Click the Number Plan tab. Expand the folders listed in the file tree to view all available extensions within your organizations Sphere system. Highlight the voice mail extension you wish to configure. Click Properties. Select Round Robin from the Hunt Order list. Click OK.
Repeat this process to configure the hunt order for your organizations auto attendant extension.
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Click the User Rights tab. Expand the folders listed in the file tree to view all available user accounts within your organizations Sphere system. Highlight the SMDI instance. Click Properties.
Click OK.
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Media servers within a Sphere system define the MSRIDs and MSRLineIDs required for integration with a voice mail server. Configuration settings defined within each media server must mirror the appropriate, corresponding settings on the voice messaging server.
Click the General tab. Select Media Servers from the list. Right-click and select Add.
Type the name of this Media Server in the Name field. Note: You must configure a unique message store for each voice messaging server
Type the name of the Sphericall Manager running the SMDI process in the Server field.
The name in this field may or may not be the name of the server upon which voice mail is installed. However, at least one Sphericall Manager throughout your organization must run the SMDI process if you wish to integrate the Sphere system with voice mail.
6
This is a unique identifier that designates a particular Message Server Retrieval system. The number (between 000 and 999) is prepended with zeros to pad the length. This entry is typically 001 and is matched on the voice mail platform.
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Select the COM port that will be connecting the Sphericall Manager to the voice mail server from the Serial Port list. Note: The specified COM port must be the COM port on the server that is running
Select the number of digits (7 or 10) that the MSR is expecting to receive from the Number of Digits list.
This 7- or 10-digit number is the caller ID information sent to the voice mail server. This number should be consistent with the settings in the Properties for Trunk window\Inward Routing tab and the settings on the voice mail server.
Note: Sphere Communications recommends that you select 10 (which equates to
the area code plus the 7-digit telephone number) as the caller ID digit length.
9
Select how unknown numbers are to appear as caller ID information from the Unknown Number list.
Select Blank if caller ID information is to display nothing when no caller ID information is available. Select Zero if caller ID information is to display all zeros when no caller ID information is available.
Note: Sphere Communications recommends that you select Zero from the Unknown
Number list. The presence or absence of these zero digits aids in the verification of message receipt during troubleshooting scenarios. In the Lines area:
10
These stations connect directly from the PhoneHub and/or BranchHub ports to the voice mail servers Dialogic card(s). These stations are those that were associated with the phantom extensions you created for voice mail functionality.
11 12 13
Expand the MG(s) listed in the display area to view all available stations within your organizations Sphere system. Highlight the station(s) you wish to associate with this MSR. Click OK.
In the MSR Line ID column, type the appropriate MSR Line ID, in the appropriate field, for each line assignment.
The MSR Line ID is an integer prepended with zeroes to create a 4-digit integer. It identifies the appropriate station to the associated message store.
Note: The MSR Line ID must match the associated port number on the voice mail
server.
CAUTION! ALTHOUGH THE MSR LINE ID DOES NOT HAVE TO BEGIN WITH 0001 AND INCREMENT BY 1, A DIRECT CORRELATION MUST EXIST BETWEEN THE SETTINGS ON THE SPHERE SYSTEM AND THE SETTINGS ON THE VOICE MESSAGING PLATFORM. SPHERE COMMUNICATIONS STRONGLY RECOMMENDS THAT, FOR EASE OF INSTALLATION AND ADMINISTRATION, THE MSR LINE ID SERIES BEGINS WITH 0001 AND INCREMENTS BY 1 FOR EACH SUBSEQUENT LINE.
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Click OK.
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User information may be exported from the Sphere system to be imported the SMDI connect voice messaging system. The information is delivered in a CSV file.
To export to CSV
On the Sphericall Administrator application:
1 2 3 4 5 6 7
Select File\Export. In the Export field, drop down to select Media Server CSV. Choose the correct Zone and/or Media Server. Optionally you may choose Template Mailbox. Select the name order for your exported data: Lastname, Firstname or Fristname Lastname. Click Export. Enter the folder location and name the file. Copy the file and load on the Media Server of the 3rd party voice messaging server.
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Once the integration of the Sphere system and voice mail is complete, calls can be sent from the addresses within the Sphere system to the appropriate ports on the voice mail server in order to verify connection status. If your organizations integration has been designed with breakout boxes and RJ-11 cables, you can test all of the voice messaging port assignments in this manner. When you are finished with the testing, you should replace all RJ-11 cables and ensure full connection between the Sphere system and the voice mail server. To complete a full range of testing for the integration, you should verify the functionality of the following features: Direct subscriber access of mailboxes Direct calls into voice mail by the operator Set MWI and Cancel MWI (with MWI-capable telephones only) Call forward to a personal greeting Call disconnect
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Verify that the Sphere system is programmed appropriately to pass integration information to voice mail.
tab.
d. Select None from the Subscriber Mailbox Options window\Answering tab\Busy
Open the Sphericall Administrator application. Assign the appropriate user rights to the operator in the User Rights tab.
Open the Sphericall Desktop. Open the appropriate line to be used by the operator.
From the Sphericall Desktop window (open on the operators client workstation):
7
The caller whom the operator wished to transfer is now connected, after a 2-second delay, to voice mail.
This test involves the use of two test telephones. From the first test telephone:
1
Enter the appropriate security code to access the Subscriber mailbox. Record a message to be delivered to the first test Subscriber mailbox.
Place a call to the lead number of the SMDI integration. Enter the appropriate security code to access the Subscriber mailbox. Listen to the test message. Delete the message.
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Confirm that the MWI has been canceled on the test telephone.
Manually cancel MWI for the affected station by using the Administrative Star Code *9710 to disable the MWI.
Note: if a user has more than one extension assigned to that station, the MWI may continue to flash until all the extension messages have been received.
Open the Sphericall Administrator application. Click the Number Plan tab. Configure fowarding conditions for the test telephone stations so that calls will be covered to voice mail after 3 rings.
With coverage enabled, the Sphericall Manager will send any calls to your extension(s) that remain unanswered after 3 rings to voice mail. From the first test telephone:
4 5
Place a call to the second test telephone. Do not answer the test call.
After 3 rings, the Sphericall Manager should send the call to voice mail.
6
Use the Monitor status utility to verify that voice mail completes the following actions:
a. Answers the call sent from the Sphericall Manager. b. Forwards it to the appropriate Subscriber mailbox.
Once voice mail answers the call, you should hear the personal greeting recorded for this Subscriber mailbox.
7 8 9
Record a message for this mailbox. Hang up the telephone. Use the Monitor utility to verify that voice mail disconnects the call once it is finished processing the recorded message for the appropriate subscriber.
Dial the extension of the voice messaging system. Record name(s) and personal greeting(s) for a sampling of the voice mailboxes and their corresponding extensions. Dial one (ore more) of the configured-for-voice mail extensions and listen to the greeting. Leave a voice message to the extensions voice mailbox. Verify that the MWI light flashes on the recipients telephone set once you are finished leaving the sample voice message. Dial the extension of the voice messaging system. Enter the appropriate security code for the intended recipients extension/voice mailbox. Listen to the voice message. Forward the voice message to another voice mailbox on the system. Verify that the MWI light flashes on the second recipients telephone set once you are finished with forwarding the sample message. Delete the voice message.
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12
Verify that the MWI light turns off on the recipients telephone.
Dial the extension of the auto attendant for the system. Verify your organizations greeting(s). Dial the extension of the auto attendant. Dial 0 (or the appropriate operator extension) to test utilization of the auto attendant to reach the operator.
Description
Message desk (incoming call record) Message waiting indication (error record) Message desk number or MSRID Message desk terminal number or MSR Line ID Any of the following letters where: A = forwarded all calls ("is unavailable") N = forwarded on ring-no-answer ("doesnt answer") B = forwarded on busy D = direct call to the voice messaging platform ("to enter your mailbox") U = unknown reason code (call sent to auto attendant) Called party station number followed by a space. If no called party, then just a space Calling party station number (if known) followed by a space. If no called party, then just a space (depending on voice application platform. If the number is 7 digits and the platform requires 10 digits, it will prepend the number with zeros. Indicates a space separating information.
xxxxxxx yyyyyyy
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yyyyyyyyyy is the calling station number. The calling station may be omitted, in which case xxxxxxxxxx is replaced with a <sp>. Calling and forwarding number will be either 7 or 10 digits. Forward On Unknown Reason <cr><lf>MDgggmmmmUxxxxxxxxxx<sp>yyyyyyyyyy<sp><cr><lf><ctrl y> ggg is the Message Desk Number or MSRID. mmmm is the Message Desk Terminal (MSR Line ID). xxxxxxxxxx is the forwarding from station or called station number. yyyyyyyyyy is the calling station number. The calling station may be omitted, in which case xxxxxxxxxx is replaced with a <sp>. Calling and forwarding number will be either 7 or 10 digits. This message is typically used to trigger the auto attendant menu linked to the called party, xxxxxxxxxx. Invalid Message Waiting <cr><lf>MWIxxxxxxxxxx<sp>INV<cr><lf><ctrl y> xxxxxxxxxx is the forwarding from station or called station number. This is the mailbox to which the voice messaging system will link this message. Calling and forwarding number will be either 7 or 10 digits. This message is sent to the voice messaging system when an MWI light request is sent for an extension that does not exist. This causes the voice messaging platform to cease transmitting the messages for this mailbox.
Note: This is currently not implemented in the Sphere system as it causes no harm
with our current loading. Blocked Message Waiting <cr><lf>MWIxxxxxxxxxx<sp>BLK<cr><lf><ctrl y> xxxxxxxxxx is the Forwarding From station or called station number. This is the mailbox to which the voice messaging system will link this message. Forwarding number will be either 7 or 10 digits. This message is sent to the voice messaging system when a message waiting light request could not be acted upon. This is currently not implemented in the Sphere system. All message waiting requests will be acted upon if received in the Sphere system. Testing is underway to see if this can be used to sync the message waiting lights on Sphericall Manager re-initializations. Message Waiting Indicator Messages OP:MWI<sp>xxxxxxxxxx!<ctrl D> Turns on (operates) the message waiting light for the extension at xxxxxxxxxx. RMV:MWI<sp>xxxxxxxxxx!<ctrl D>
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Turns off (removes) the message waiting light for the extension at xxxxxxxxxx.
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Verify a secure connection at the Sphericall Manager(s). Verify a secure connection at the voice messaging platform(s).
Resolve serial cable or COM port operability issues within the Sphericall/SMDI integration.
HYPERTERMINAL MONITORING
Sphere Communications recommends the use of HyperTerminal to monitor the status of information processed through the appropriate COM port(s) within a Sphericall/SMDI integration.
Click Start\Programs\Accessories\Communications\HyperTerminal.
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Figure 11.1
2 3 4
Type the name of this connection in the Name field. Select an icon for the connection. Click OK.
Figure 11.2
Connect To window
5 6
Select the appropriate COM port for connection to the voice messaging platform from the Connect using list. Click OK.
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Figure 11.3
7 8
Configure the COM port as necessary for the appropriate connection to the voice messaging platform. Click OK.
Figure 11.4
HyperTerminal window
You can now view the SMDI packets as they pass through the selected COM port.
Detach the RS-232 cable from the voice mail servers serial port.
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Connect the cable to another COM port on the Sphericall Manager or on another network resource in order to place the Sphericall Manager into loopback state. or Generate calls to voice mail. Monitor the HyperTerminal utility to verify that the Sphericall Manager is attempting to send information to the voice mail server. Reconnect the RS-232 (Null Modem) cable to the voice messaging platform.
3 4
Dial an extension within your organizations Sphericall numbering plan. Leave a voice message in the Subscriber mailbox associated with that extension. Monitor the appropriate station to ensure that the telephones MWI light flashes. or Monitor the appropriate station to ensure that the stutter dial tone sounds in the telephone handset, indicating a waiting voice message.
Auto attendant and voice mail do not answer. Also, a power outage had occurred and both servers were at the NT logon.
Q3870
Users receive an error message each time they attempt to access voice mail.
Q3838
Callers hear the Name Not Recognized message when they enter letters of a subscriber's last name within the company directory. Mailboxes are set up and names have been recorded.
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Sphericall and voice mail systems has been compromised. Issues of this sort are, at times, difficult to detect in that the Sphericall system will continue to function normally even though voice messaging sessions are never offered to calls within the integration. Sphere recommends the use of the Sphere Troubleshooting Methodology to ascertain the source of any problems in the system. This problem solving methodology is found in the Sphericall Maintenance and Troubleshooting Manual.
Verify a secure connection at the Sphericall Manager(s). Verify a secure connection at the voice messaging platform(s).
Resolve serial cable or COM port operability issues within the Sphericall/SMDI integration.
HYPERTERMINAL MONITORING
Sphere Communications recommends the use of HyperTerminal to monitor the status of information processed through the appropriate COM port(s) within a Sphericall/SMDI integration.
Click Start\Programs\Accessories\Communications\HyperTerminal.
Figure 11.5
2 3 4
Type the name of this connection in the Name field. Select an icon for the connection. Click OK.
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Figure 11.6
Connect To window
5 6
Select the appropriate COM port for connection to the voice messaging platform from the Connect using list. Click OK.
Figure 11.7
7 8
Configure the COM port as necessary for the appropriate connection to the voice messaging platform. Click OK.
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Figure 11.8
HyperTerminal window
You can now view the SMDI packets as they pass through the selected COM port.
Detach the RS-232 cable from the voice mail servers serial port. Connect the cable to another COM port on the Sphericall Manager or on another network resource in order to place the Sphericall Manager into loopback state. or Generate calls to voice mail. Monitor the HyperTerminal utility to verify that the Sphericall Manager is attempting to send information to the voice mail server. Reconnect the RS-232 (Null Modem) cable to the voice messaging platform.
3 4
Dial an extension within your organizations Sphericall numbering plan. Leave a voice message in the Subscriber mailbox associated with that extension. Monitor the appropriate station to ensure that the telephones MWI light flashes. or Monitor the appropriate station to ensure that the stutter dial tone sounds in the telephone handset, indicating a waiting voice message.
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Auto attendant and voice mail do not answer. Also, a power outage had occurred and both servers were at the NT logon.
Q3870
Users receive an error message each time they attempt to access voice mail.
Q3838
Callers hear the Name Not Recognized message when they enter letters of a subscriber's last name within the company directory. Mailboxes are set up and names have been recorded.
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Sphericall Manager to the Exchange server and receive a reply, and you must be able to do this in reverse order as well. The RPC service must be running. RPC calls are used for a majority of TAPI and MAPI functions as well as some functionality from the vmail.exe process to the pbx.exe process. If all of these factors are functioning as specified, the voice mail service will work.
Possible Cause
The serial packets are not reaching the voice mail server or the packets are not 100% accurate. This can be caused by an incorrect cable, non-matching serial port configurations, or incorrect serial port usage. Whenever a voice mail system has received no packet or a packet that it does not recognize, it will tend to wait some time while it attempts to retrieve a recognizable packet. If it does not receive this packet, it will time out and travel to the main auto attendant. If the cable is plugged in to an incorrect serial port in the back of either machine, the packet will not be recognized by the voice mail system.The communication port on your organizations server might not be set appropriately. Each of the serial ports on the motherboard has the capability to be any of four different communication ports. It is frequently assumed that COM1 is the topmost port on the back of the server. This is not always the case. Note: The system BIOS provides port settings. Upon reinitializing the server, press the delete key or F2 to enter the system setup. Under the integrated peripherals option you can see the com port setting for the serial ports on the motherboard. The first serial port is typically the top or the furthest to the left. These settings relate to the operating system view of the ports: COM1: Address 3f8 IRQ 4 COM2: Address 2f8 IRQ 3 COM3: Address 3e8 IRQ 4 COM4: Address 2e8 IRQ 3
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Problem A caller accesses the wrong mailbox when trying to leave a message.
Symptom
After a caller is transferred to the voice mail system from coverage conditions, the caller is given the wrong voice mailbox greeting from the person with whom they wished to speak.
Possible Cause
The MSR Line ID is not configured with the correct values. The MSR Line ID is a value that corresponds to the actual physical line number that is entering the voice mail server. The field is a 4-digit value that is pre-pended with zeroes. For example, the first line into the voice mailbox would be 0001.This number has to match up with the line ID on the voice mail server; it is standard to make the first line 0001, the second line 0002, etc. This line identifier is sent over the serial port with the call packet to tell the voice mail that the call that is coming over line one has X packet. The system will see a phone call coming in on a specific line and wait for the related packet until it answers the call. If two lines are switched and they each get a call, both callers will get an incorrect voice mailbox. The MSR Line ID is not configured with the correct value on the last few lines. This will only show itself during peak calling hours if the hunt group is set to Linear. If the last few lines are reversed, you may either get an incorrect voice mailbox or the system will time out to the auto attendant.
After a caller is transferred to the voicemail system from coverage conditions, the caller is given the wrong voice mailbox greeting from the person with whom they wished to speak. The error message SerialPort Open Failed for COM1: appears.
The serial port is no longer available from the system. Microsoft Windows NT does not allow programs to share system devices such as parallel ports. If the device is used by another application (such as a modem) when SMDI has started, this error message will appear. Either choose a different serial port or remove the other user of the appropriate serial port. The affected line does not have stutter dial tone enabled. From the Sphericall Administration application, you must enable stutter dial tone under the Properties For Multiple Stations window.
Users receive no stutter dial tone on a phone without a message waiting light.
Users receive no stutter dial tone on a phone without a message waiting light; they just hear a crackling every half second. Users receive no message waiting indication at all.
User perception of stutter dial tone is incorrect. Most people expect to hear what is called Message Waiting dial tone so they are usually confused by true stutter dial tone. Change the system setting to message waiting dial tone.
The cable is only a simplex cable with one pair from Transmit on the PBX side to Receive on the voice mail side. Change the cable to a duplex cable. The voice mail system is not set to send message waiting notification. This is typically a configured option and will need to be set correctly on the voice mail side. Another possible cause is incorrect serial port configuration. Certain combinations allow one-way communications to have few errors while the other way might have more. Confirm that the configurations are the same.
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Summary
Symptom
When a user dials in from the outside, he hears the auto attendant but is unable to retrieve voice mail.
Possible Cause
When integrated with Microsoft Exchange Server 5.0, name resolution is faulty. If the Sphericall Softswitch can not resolve the name of Exchange Server, the auto attendant will still answer calls and take voice mail messages even though you cannot retrieve them. This is because all greetings are cached on the Sphericall Softswitch after first connection to Microsoft Exchange Server 5.0. If name resolution changes between the time the Sphericall Softswitch first starts and the attempt to retrieve messages, voice mail messages might be unretrievable through the telephone user interface. When integrated with Microsoft Exchange Server 5.0, the mailbox is no longer associated with the extension. This is possibly caused by the Administrator removing the mailbox from the extension in the Sphericall Administration application. Dialed phone number incorrectly (user error). The user is entering an invalid password. Change the password from Microsoft Outlook 98.
When a user dials in from the outside he is able to attempt voice mail dialing but is told that he has entered an invalid extension.
The voice mail auto attendant does not answer at the main number. Greetings are known to have been changed but have not taken effect. The user is unable to transfer to voice mail or reach the auto attendant.
The old greeting is playing after the greeting was changed. The user dials the main number but gets a busy tone.
There may be a timer for refreshing the greetings. If you change your voice mail greetings, allow approximately five minutes for the system to update changes before confirming changes. Voice mail ports may be busy. This will happen at peak times of the day when everyone is checking voice mail from the phone or when customers are calling into the system. This usually occurs around an organizations opening and closing time.
SUMMARY
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Upon completing this chapter, you should now have a better understanding of SMDI. You will also increase your ability to recognize voice mail issues and how to resolve them. The Sphericall system integrates with voice mail platforms utilizing SMDI, an industrystandard method of integrating a voice messaging platform with a PBX using the RS232c serial connection found on any personal computer.
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Summary
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Have you installed, configured, and ensured network connectivity to all of the appropriate Sphere MGs and IP phones for your organizations Sphere system? Have you configured the general System properties and functionality for your organizations Sphere system? Have you defined the numbering plan for your organizations Sphere system? Have you defined all of the global and local system settings for the Sphericall Manager? Have you installed, configured, and integrated the Sphericall Desktop on workstations throughout your organizations Sphere system?
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OVERVIEW
Administrators have the ability to monitor IP network information throughout a Sphere system via the Simple Network Management Protocol, a software-based management service that accepts information messages from endpoints using UDP. This information is essentially the same information that can be obtained via tell commands. With SNMP, you can now use an SNMP manager (in conjunction with point-and-click functionality) to gather the same information formerly obtained by using the MG command line interface. SNMP relies upon the functionality of two types of devices within a network, an SNMP manager and the SNMP agents. The SNMP manager resides on a network resource and serves as the collection point for all data generated by the traps. An SNMP managers main functions are as follows: To gather information from SNMP agents using the Get and Get-Next requests To modify settings on the appropriate SNMP agents using the Set request To accept traps from SNMP agents
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The SNMP agents are client machines or devices such as workstations, network hubs, resource servers, etc. located throughout an organizations network. All Sphere MGs are built with the SNMP agent installed. Once the SNMP agents have been configured, they are responsible for the following tasks: To respond to queries from the SNMP manager To monitor hardware devices for any service faults and notify the SNMP manager if a fault is detected
Note: Information that is passed from the SNMP agent to the SNMP manager is
called a trap (i.e. a specific error condition). You must configure each device to send traps to the SNMP manager via a trap destination address. This is a System Initialization Setting configured via the Sphericall Administrator application (SNMP Trap Destination setting). An organization may decide to use the SNMP functionality upon initial installation of the Sphere system. Sphericalls Management Information Bases must be installed on the designated SNMP manager utilizing standard installation mechanisms. Once administrators specify the location of the Sphericall MIBs, which reside on the Sphericall software DVD-ROM, they can use the SNMP manager to query Sphere hardware by selecting the appropriate OID defined in one of the MIBs.
REQUIREMENTS
The Sphere system adheres to the following SNMP standards and compatibility: SNMP Version: MIBs-II Version 1 SNMP Specification: RFC 1213, as stated in IETF definitions
Open a serial connection or a Telnet connection to the MG. Enter the following command:
password. It adds an additional password which can be used to gain access to the MG. You can also restrict access by entering the IP address of the only device to have access to the MG.
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To view all the SNMP community names, use the snmp access list command. To remove a community name from the access list, use the snmp access delete <community> command. To remove all community names, use the snmp access flush command.
MIB
Agent Software
Network
Network
Network
MIBs store all available OIDs. The administrator selects an OID and specifies the device to monitor by its IP address. The SNMP manager builds the SNMP request. The SNMP manager transmits the request to the selected device. The SNMP agent responds to the request. The SNMP manager displays the information received from the SNMP agent.
An event occurs at the SNMP agent. The SNMP agent sends a trap, as defined in the MIBs, if the event falls within a predefined set of possible events. The SNMP manager receives the SNMP agents trap. The SNMP manager references the MIBs. The SNMP manager displays the trap with other error condition severity information.
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CRITICAL TRAP INFORMATION
Refer to the following table for the specific trap information relative to a Sphere system.
Table 12.1 Sphere SNMP Traps
Sphere SNMP Traps
-24 volt supply out of specification -12 volt supply out of specification - VBX only +2.5 volt supply out of specification +3.5 volt supply out of specification +5 volt supply out of specification +12 volt supply out of specification DSP programming failed (per board basis) Fan fault detected FPGA programming failed (per board basis) HV supply voltage out of specification on board 0 HV supply voltage out of specification on board 1 Media stream connection failures (per board basis) MGC connection lost MGC connection gained Temperature too high on engine board Temperature too high on LM78 - VBX only Trunk came up Trunk went down Unknown board type detected (per board basis)
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OVERVIEW
Once you have installed and configured the SNMP manager on a network resource server within your organizations existing network, you must load the SNMP manager with the appropriate Sphere SNMP MIBs. These MIBs can be found on the Sphericall software DVD-ROM in the \server\data\vbx folder.
INSTALLATION
Note: Because several SNMP manager applications currently exist, the following
information regarding the installation of Sphere SNMP MIBs is rather general in nature. You should integrate the Sphere SNMP MIB files into the SNMP manager according to the SNMP managers application installation instructions.
REQUIREMENTS
You must load the following two files before proceeding with the installation of the remaining Sphere SNMP MIBs: sphere-reg.mib sphere-tc.mib
Open the SNMP manager software. Load the sphere-reg.mib file. Load the sphere-tc.mib file. Load the following MIBs to complete the Sphere SNMP MIBs installation procedure:
sphere-board.mib sphere-board-trap.mib sphere-channel-trap.mib sphere-channel.mib sphere-e2prom.mib sphere-hlog.mib sphere-lm78.mib sphere-lm78-trap.mib sphere-lm80.mib (new with VG3) sphere-lm80-trap.mib (new with VG3) sphere-mgcfinder-trap.mib sphere-system.mib sphere-version.mib
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CONTENTS
Refer to the following table for a list of the information provided and available within each Sphere SNMP MIB.
Table 12.2 SNMP SNMP MIBs Contents
Name Table of version objects
Contents
Telephony subsystem build date and time Telephony controller version Media stream version Station version Trunk version Telephony driver version Multi-media version MGC interface version CPLD version - VBX units only Amount of free memory Port type MGC address Number of installed channels Number of connected channels Loop current loss threshold Number of active calls Total number of calls Board type Board ID Board revision DSP bootstrap image description, date, and time DSP image description, date, and time FPGA image description, date, and time Echo canceller type Line termination impedance Channel installed status MGC connection status Hook state status Call in progress status Volume setting Gain setting
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Contents
60 volt sense reading for board 0 60 volt sense reading for board 1 5 volt supply reading - VBX units only 12 volt supply reading -12 volt supply reading -24 volt supply reading Lm78 chip temperature
VG3 Units: -58 volt sense reading for board 0 -58 volt sense reading for board 1 +2.5 volt supply reading +3.3 volt supply reading +5 volt supply reading +12 volt supply reading +24 volt supply reading
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Object IDs comprise the base of data within Spheres SNMP MIBs. Refer to the following table for an index of the Sphere SNMP OIDs that are able to be monitored via SNMP within a Sphere system.
Table 12.3
OID Name
sphereRoot sphereRegistration spherePlatformReg sphereVimReg sphereHubReg sphereCohubReg spherePhonePortReg sphereBranchReg sphereSystemsReg sphereGeneric spherePlatformGen spherePfmGenObjs historicLogTable
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OID Name
historicLogEntry sequenceNumber notificationId notificationNumber affectedEntityId severity timestamp ipAddress macAddress phoneNumber dataId versionTable versionEntry versionIndex versionTelBuildInfo versionTelController versionMediaStream versionStation versionTrunk versionTelDriver versionMultiMedia versionMgcInterface versionCpld systemTable systemEntry systemIndex systemFreeMemory systemSyncSource systemPortType systemMgcAddress systemInstalledChannels systemConnectedChannels
OID Number
1.3.6.1.4.1.4613.2.1.1.1.1 1.3.6.1.4.1.4613.2.1.1.1.1.1 1.3.6.1.4.1.4613.2.1.1.1.1.2 1.3.6.1.4.1.4613.2.1.1.1.1.3 1.3.6.1.4.1.4613.2.1.1.1.1.4 1.3.6.1.4.1.4613.2.1.1.1.1.5 1.3.6.1.4.1.4613.2.1.1.1.1.6 1.3.6.1.4.1.4613.2.1.1.1.1.7 1.3.6.1.4.1.4613.2.1.1.1.1.8 1.3.6.1.4.1.4613.2.1.1.1.1.9 1.3.6.1.4.1.4613.2.1.1.1.1.10 1.3.6.1.4.1.4613.2.1.1.2 1.3.6.1.4.1.4613.2.1.1.2.1 1.3.6.1.4.1.4613.2.1.1.2.1.1 1.3.6.1.4.1.4613.2.1.1.2.1.2 1.3.6.1.4.1.4613.2.1.1.2.1.3 1.3.6.1.4.1.4613.2.1.1.2.1.4 1.3.6.1.4.1.4613.2.1.1.2.1.5 1.3.6.1.4.1.4613.2.1.1.2.1.6 1.3.6.1.4.1.4613.2.1.1.2.1.7 1.3.6.1.4.1.4613.2.1.1.2.1.8 1.3.6.1.4.1.4613.2.1.1.2.1.9 1.3.6.1.4.1.4613.2.1.1.2.1.10 1.3.6.1.4.1.4613.2.1.1.3 1.3.6.1.4.1.4613.2.1.1.3.1 1.3.6.1.4.1.4613.2.1.1.3.1.1 1.3.6.1.4.1.4613.2.1.1.3.1.2 1.3.6.1.4.1.4613.2.1.1.3.1.3 1.3.6.1.4.1.4613.2.1.1.3.1.4 1.3.6.1.4.1.4613.2.1.1.3.1.5 1.3.6.1.4.1.4613.2.1.1.3.1.6 1.3.6.1.4.1.4613.2.1.1.3.1.7 N/A N/A N/A N/A N/A N/A N/A N/A N/A N/A N/A N/A N/A N/A telcon info telcon info telcon info telcon info telcon info telcon info telcon info telcon info engine hw N/A N/A N/A chips mem N/A N/A
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OID Name
systemLclThreshold systemVoiceOnThreshold systemVoiceOffThreshold systemHysteresisTimer systemActiveCalls systemTotalCalls boardTable boardEntry boardIndex boardType boardId boardRevision boardDspBootstrapDesc boardDspImageDesc boardFpgaImageDesc boardEchoCancellerType boardLineTermImpedance channelTable channelEntry channelIndex channelInstalled channelConnectedToMgc channelHookState channelCallInProgress channelVolumeSetting channelGainSetting e2promTable e2promEntry e2promIndex e2promScrollRate e2promPeakCellRate e2promTempUnits
OID Number
1.3.6.1.4.1.4613.2.1.1.3.1.8 1.3.6.1.4.1.4613.2.1.1.3.1.9 1.3.6.1.4.1.4613.2.1.1.3.1.10 1.3.6.1.4.1.4613.2.1.1.3.1.11 1.3.6.1.4.1.4613.2.1.1.3.1.12 1.3.6.1.4.1.4613.2.1.1.3.1.13 1.3.6.1.4.1.4613.2.1.1.4 1.3.6.1.4.1.4613.2.1.1.4.1 1.3.6.1.4.1.4613.2.1.1.4.1.1 1.3.6.1.4.1.4613.2.1.1.4.1.2 1.3.6.1.4.1.4613.2.1.1.4.1.3 1.3.6.1.4.1.4613.2.1.1.4.1.4 1.3.6.1.4.1.4613.2.1.1.4.1.5 1.3.6.1.4.1.4613.2.1.1.4.1.6 1.3.6.1.4.1.4613.2.1.1.4.1.7 1.3.6.1.4.1.4613.2.1.1.4.1.8 1.3.6.1.4.1.4613.2.1.1.4.1.9 1.3.6.1.4.1.4613.2.1.1.5 1.3.6.1.4.1.4613.2.1.1.5.1 1.3.6.1.4.1.4613.2.1.1.5.1.1 1.3.6.1.4.1.4613.2.1.1.5.1.2 1.3.6.1.4.1.4613.2.1.1.5.1.3 1.3.6.1.4.1.4613.2.1.1.5.1.4 1.3.6.1.4.1.4613.2.1.1.5.1.5 1.3.6.1.4.1.4613.2.1.1.5.1.6 1.3.6.1.4.1.4613.2.1.1.5.1.7 1.3.6.1.4.1.4613.2.1.1.6 1.3.6.1.4.1.4613.2.1.1.6.1 1.3.6.1.4.1.4613.2.1.1.6.1.1 1.3.6.1.4.1.4613.2.1.1.6.1.2 1.3.6.1.4.1.4613.2.1.1.6.1.3 1.3.6.1.4.1.4613.2.1.1.6.1.4
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OID Name
e2promVpiNumber e2promWatchdogCycles lm78Table lm78Entry lm78Index lm78Neg60VoltBoard0 lm78Neg60VoltBoard1 lm78Pos5VoltSupply lm78Pos12VoltSupply lm78Neg12VoltSupply lm78Neg24VoltSupply lm78ChipTemperature Im80Table Im80Entry Im80Index Im80Neg58VoltBoard0 Im80Neg58VoltBoard1 Im80Pos2Pt5VoltBoard Im80Pos3Pt3VoltBoard Im80Pos5VoltSupply Im80Pos12VoltSupply Im80Neg24VoltSupply spherePfmGenTraps spherePfmBoardTraps boardTypeTrap boardDspProgramTrap boardFpgaProgramTrap spherePfmChannelTraps channelMsConnectionTrap spherePfmLm78Traps lm78TempHighEngTrap lm78Neg60vBoard0Trap
OID Number
1.3.6.1.4.1.4613.2.1.1.6.1.5 1.3.6.1.4.1.4613.2.1.1.6.1.6 1.3.6.1.4.1.4613.2.1.1.7 1.3.6.1.4.1.4613.2.1.1.7.1 1.3.6.1.4.1.4613.2.1.1.7.1.1 1.3.6.1.4.1.4613.2.1.1.7.1.2 1.3.6.1.4.1.4613.2.1.1.7.1.3 1.3.6.1.4.1.4613.2.1.1.7.1.4 1.3.6.1.4.1.4613.2.1.1.7.1.5 1.3.6.1.4.1.4613.2.1.1.7.1.6 1.3.6.1.4.1.4613.2.1.1.7.1.7 1.3.6.1.4.1.4613.2.1.1.7.1.8 1.3.6.1.4.1.4613.2.1.1.8 1.3.6.1.4.1.4613.2.1.1.8.1 1.3.6.1.4.1.4613.2.1.1.8.1.1 1.3.6.1.4.1.4613.2.1.1.8.1.2 1.3.6.1.4.1.4613.2.1.1.8.1.3 1.3.6.1.4.1.4613.2.1.1.8.1.4 1.3.6.1.4.1.4613.2.1.1.8.1.5 1.3.6.1.4.1.4613.2.1.1.8.1.6 1.3.6.1.4.1.4613.2.1.1.8.1.7 1.3.6.1.4.1.4613.2.1.1.8.1.8 1.3.6.1.4.1.4613.2.1.2 1.3.6.1.4.1.4613.2.1.2.1 1.3.6.1.4.1.4613.2.1.2.1.1 1.3.6.1.4.1.4613.2.1.2.1.2 1.3.6.1.4.1.4613.2.1.2.1.3 1.3.6.1.4.1.4613.2.1.2.2 1.3.6.1.4.1.4613.2.1.2.2.1 1.3.6.1.4.1.4613.2.1.2.3 1.3.6.1.4.1.4613.2.1.2.3.1 1.3.6.1.4.1.4613.2.1.2.3.2 N/A N/A N/A N/A N/A
engine list menus engine list menus engine list menus engine list menus engine list menus engine list menus engine list menus engine list menus engine list menus engine list menus engine list menus engine list menus engine list menus engine list menus engine list menus engine list menus engine list menus N/A N/A N/A N/A N/A N/A N/A N/A N/A N/A
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SNMP INTEGRATION
OID Name
lm78Neg60vBoard1Trap lm78Pos5vSupplyTrap lm78TempHighChipTrap lm78Pos12vSupplyTrap lm78Neg12vSupplyTrap lm78Neg24vSupplyTrap lm78FanFaultTrap spherePfmMgcFinderTraps mgcFinderConLostTrap
OID Number
1.3.6.1.4.1.4613.2.1.2.3.3 1.3.6.1.4.1.4613.2.1.2.3.4 1.3.6.1.4.1.4613.2.1.2.3.5 1.3.6.1.4.1.4613.2.1.2.3.6 1.3.6.1.4.1.4613.2.1.2.3.7 1.3.6.1.4.1.4613.2.1.2.3.8 1.3.6.1.4.1.4613.2.1.2.3.9 1.3.6.1.4.1.4613.2.1.2.4 1.3.6.1.4.1.4613.2.1.2.4.1 N/A N/A N/A N/A N/A N/A N/A N/A N/A
Note: This trap is network connection-based, not channel-based. For example, if a Sphere PhoneHub loses its connection to a Sphericall Manager, only one trap will be sent to the SNMP manager from the SNMP agent rather than one trap for each affected channel on that MG. mgcFinderConGainTrap 1.3.6.1.4.1.4613.2.1.2.4.2 N/A
Note: This trap is network connection-based, not channel-based. For example, if a Sphere PhoneHub loses its connection to a Sphericall Manager, only one trap will be sent to the SNMP manager from the SNMP agent rather than one trap for each affected channel on that MG. mgcFinderTrunkDownTrap mgcFinderTrunkUpTrap sphereSystemsGen sphereSysGenObjs sphereSysGenTraps sphereProduct spherePlatformProd sphereSystemsProd sphereCapabilities sphereRequirements sphereExperimental 1.3.6.1.4.1.4613.2.1.2.4.3 1.3.6.1.4.1.4613.2.1.2.4.4 1.3.6.1.4.1.4613.2.2 1.3.6.1.4.1.4613.2.2.1 1.3.6.1.4.1.4613.2.2.2 1.3.6.1.4.1.4613.3 1.3.6.1.4.1.4613.3.1 1.3.6.1.4.1.4613.3.2 1.3.6.1.4.1.4613.4 1.3.6.1.4.1.4613.5 1.3.6.1.4.1.4613.6 N/A N/A N/A N/A N/A N/A N/A N/A N/A N/A N/A
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SNMP INTEGRATION
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13
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Previously, Sphericall Desktops were installed using either an installation CD/DVD or shared network drive containing a copy of the CD/DVD. Installers were required to have administrative privileges for the PC on which the Sphericall Desktop is being installed. The initial installation of larger Sphere systems required several installers to walk around logging into client PCs and loading the Sphericall Desktop. This process is time consuming. Sphericall introduces support for Microsoft Windows Installer. Windows installation packages can be managed using the Active Directorys Group Policy thereby centralizing installation.
Note: Sphere Communications assumes that the Sphere system administrator has
experience with Microsoft networking and Group Policy. Microsoft Windows Installer is an installation and configuration service that reduces the total cost of ownership. The Installer ships with Windows Vista, the Windows Server 2003 family, Windows XP, and Windows 2000.
....
I.GROUP POLICY
Using Windows Installer, the Sphere administrator has a way of installing, upgrading or uninstalling the Sphericall Desktop on PCs of users who do not have local administrative privileges for their PCs. Instmsi.exe is the redistributable package for installing or upgrading Windows Installer. To manage an MSI package, use the following Group Policy: User Configuration (or Computer Configuration)->Software installation->New package. Sphericall does not provide zap installation files. Group Policy is not supported on Windows 98, ME or Windows NT Workstation 4.0. The Sphericall_Desktop.msi and its supporting files are placed into a shared directory. This directory does not have to be on a Sphericall Manager. A true file server can be used for better performance.
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administrative privileges, the following Group Policy setting should be enabled: User Configuration (or Computer Configuration)->Administrative Templates->Windows Components->Windows Installer-> "Always install with elevated privileges". B. Allowing User Input If the administrator wants users to input installation parameters as when the Maximum user interface option is used, the following Group Policy setting should be enabled: Computer Configuration->Administrative Templates->Windows Components->Windows Installer->"Enable user control over installs".
DEPLOYMENT SETTINGS
The deployment type defines how the Sphere installation process is initiated: A. Published You publish an application when you want the application to be available to people managed by the Group Policy object, should a user want the application. With published applications, it is up to each person to decide whether or not to install the published application. When the Sphericall Desktop is published, the program is available in the client's Add/Remove Programs -> Add New Programs list within the client's Control Panel. Pressing "Add" will start the Sphericall Desktop installation. B. Assigned You assign an application when you want everyone to have the application on his or her computer. When the Sphericall Desktop is assigned a Sphericall Desktop icon will appear in each user's Start menu. The first time the application is invoked, the installation process will begin. C. Run at Logon This setting allows the installation to run automatically when the user logs onto the system.
B. Maximum The "Installation User Interface Option" of "Maximum" allows the user the opportunity to set the install path, language and primary server name. The Maximum option should not be used with the AutoInstall.ini configuration file because the user-entered values will be overridden by values from the file. Using the Maximum user interface option presents the following dialog boxes to the user: License Agreement
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Language Selection Destination Folder Primary Sphericall Manager Ready to Install Confirmation Installation Status /Installation Complete
Table 13.1 U/I Option Comparison
No AutoInstall.ini settings
Basic User I/F Not supported There is no way to configure the server name. Supported Users are prompted for server name, language and install path.
AutoInstall.ini settings
Recommended choice Administrator defines server name & language. Install path is hard coded. Not recommended The user input will be ignored in favor of the AutoInstall.ini parameters. This may be confusing for users.
UNINSTALL
Users that have been granted permissions to uninstall software may use the Control Panels Add or Remove Programs to uninstall the Sphericall Desktop. Group Policy can be configured to uninstall the Sphericall Desktop from client PC. The uninstall must be set to run with elevated security privileges.
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COMPONENTS
Sphericall_Desktop.msi is an InstallShield built installation program. It does the bulk of the file copying and registry setup. The registry keys and values that are added are the same for each installation. InstallConfig.exe is an MFC application used for loading the TAPI service provider and for creating the more dynamic registry settings. InstallConfig.exe uses AutoInstall.ini to load administrator-defined preferences. Preferences include definition of the primary Sphericall Manager and Desktop language. AutoInstall.ini will be shipped with no preferences defined.
DISADVANTAGES
The Desktop Manager has an integrated display that offers an inventory of all the PC clients that have ever updated. Each entry includes: operating system, date of last upgrade, whether or not an upgrade is needed, and logon account of user who did the last upgrade.
Copy all files/folders to the appropriate distribution point. Based upon the distribution point, modify the AutoInstall.ini file accordingly.
The lines that have a ; at the beginning are comments and will be ignored.
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Server = must contain the primary Sphericall Manager server name Language = 0 is english (default), 1 is Mexico-Spanish, 2 is German, 3 is Italian, 4 is French-Canada, 5 is French (France), 6 is Spanish (Spain). The Sphericall Desktop MSI installation script (Sphericall Desktop.msi) will use this AutoInstall.ini file If it is located in the sub directory \Program Files\Sphere of the root directory from where the Sphericall Desktop.msi is located. AutoInstall.ini is typically used when the Sphericall Desktop is installed through group policy using the basic user interface option. The use of this file is not recommended when the Sphericall Desktop installation is run from group policy using maximum user interface option. Its also not recommended when executing Sphericall Desktop.msi directly. The field values defined here takes precedence over input provided by the user/installer.
3
Click Start\Control Panel. Click Add or Remove Programs. Click Add New Programs.
Click Add.
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A UDIOC ODES
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14
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The Sphere system should be installed, configured and tested as fully functional. Refer to the MP11XFXO Analog Gateway User Manual for installation planning, setup, package contents, safety and conditions of use.
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OVERVIEW OF OPERATION
The AudioCodes MP11X Analog Media Gateway is a multi-port analog gateway that connects analog terminals, PBXs or key systems to the IP network using FXO connectivity. Using AudioCodes Analog Media Gateways, Sphere system Administrators can effectively deliver carrier-hosted converged services as well as enterprise-based applications. MP112 - 2 Port FXO Gateway MP114 - 4 Port FXO Gateway MP118 - 8 Port FXO Gateway
Note: The AudioCodes MP11X series spans ranging from 2 to 8 analog ports. The X
identical.
Choose this method if the installer doesnt have access to the DHCP lease list.
1
Choose a computer that will be used for configuring the AudioCodes gateway. The computer will have its network disrupted during configuration so do not choose an inservice Sphericall Manager.
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AUDIOCODES
2 3 4 5 6
Make sure the AudioCodes configuration files are accessible from the configuration computer (copied onto or accessible via the Sphericall DVD). Power up the AudioCodes FXO MP11X gateway. A new AudioCodes FXO MP11X gateway uses a statically bound IP address of 10.1.10.11. Change the configuration computers subnet mask to 255.255.0.0. Change its IP address to an address on the same subnet as the AudioCodes gateway (ex. 10.1.10.12). Either connect the gateway directly to a computer using a cross-over cable OR connect both the computer and gateway together using a switch / hub. Access the gateway Quick Setup screen using a web browser such as Internet Explorer. The URL is http://10.1.10.11.
Choose this method if the installer has access to the DHCP lease list.
1 2 3
Choose a PC for configuring the AudioCodes gateway that is in the same subnet as the gateway. Make sure the AudioCodes configuration files are accessible from the configuration computer (copied onto or accessible via the Sphericall DVD). Power up the AudioCodes FXO MP11X gateway. When the AudioCodes FXO MP11X gateway is connected on a network that supports DHCP, it will be given an IP address within the DHCP address scope. Determine the gateways new IP address. The DHCP provided IP address can be determined by searching for the MAC address of the gateway within the IP address lease list of the DHCP server. The MAC address of the gateway is on a sticker located on the bottom side of the unit. Access the gateway Quick Setup screen using a web browser such as Internet Explorer. Use the IP address of the gateway as the URL (ex. http://172.16.15.35).
The gateways embedded web server will prompt for a username and password. Enter Admin for both (case sensitive). Download the software image. Go to Software Updates and browse to the AudioCodes file directory. Download MP118_SIP_F4.80A.020.001.cmp to the AudioCodes gateway. After 2 minutes, refresh the browser window. A new looking configuration pane will be displayed. From the Software Updates -> Load Auxiliary Files, load the following files:
a. INI file: SampleSIP.ini
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AUDIOCODES
IP CONFIGURATION
1 2
From the Quick Setup screen, enter either a valid static IP address & subnet mask that is appropriate for the target network. In the Default Gateway IP Address field, configure the IP address of the default gateway for the subnet..
SIP PARAMETERS
3 4 5 6
Enter the name of the AudioCodes gateway in the Gateway Name field. When working with a Proxy server, set the Working with Proxy field to Yes. Enter the Proxy Name of the primary Proxy server in the Proxy IP Address field. When no Proxy is used, the internal routing table is used to route the calls. Enter the Proxy Name in the Proxy Name field. If Proxy name is used, it replaces the Proxy IP address in all SIP messages. This means that messages are still sent to the physical Proxy IP address but the SIP URI contains the Proxy name instead. Set Enable Registration to Enable. Disable = the MediaPack does not register to a Proxy server/Registrar (default). Enable = the MediaPack registers to a Proxy server/Registrar at power up and every Registration Time seconds; The MediaPack sends a REGISTER request according to the Authentication Mode parameter.
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AUDIOCODES
Click the Reset button and click OK in the prompt; the MediaPack applies the changes and restarts.
From the Sphericall Administrator Application Trunk panel: verify the gateway has been recognized by the system. Access the Properties for Hub for the gateway. Select the Gateway Admin button. An Internet Explorer will appear and attempt to access the gateways embedded web server. If this gateway is to be configured with a secondary (redundant) MGC, go to the Protocol Management screen followed by the Protocol Definition screen and enter an MGC IP address other than the primary in the Redundant Call Agent IP field.
Open the Automatic Dialing page: Protocol Management\Endpoint Settings submenu\Automatic Dialing.
In the Destination Phone Number field for a port, enter the telephone number to dial.
Incoming calls may be directed to a specific extension by specifying a valid extension. It is possible to specify the same extension for every port (e.g. the Auto Attendant) or a different extension for each port on this page (e.g. DID lines). It is possible to use the default routing specified in the Sphericall Admininstrator application by specifyng an invalid extension for one or more of the ports on the Automatic Dialing page. For example, if the nonexistant extension "xyz" is specified on the Automatic Dialing page and the domain is "spherecom.com", when an incoming call is received, the MP-11x will send an INVITE to "xyz@spherecom.com". Since "xyz" does not exist on the system, the MGC will apply the default route to the call.
3
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AUDIOCODES
Advanced Configuration
Enable [1] When a port is selected, when making a call, the number in the Destination Phone Number field is automatically dialed if ring signal is applied to port. Disable [0] The automatic dialing option on the specific port is disabled (the number in the Destination Phone Number field is ignored). Hotline [2] When a phone is offhook and no digit is pressed for HotLineDialToneDuration, the number in the Destination Phone Number field is automatically dialed.
4 5
Repeat step 3 for each port you want to use for Automatic Dialing. Click the Submit button to save your changes.
ADVANCED CONFIGURATION
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It is strongly recommended that your organizations Sphere Administrator change these settings for proper operation of the device.
CHANNELSELECTMODE*
Menu: Protocol Management\Protocol Definitions\General Parameters Display: Channel Select Mode Recommended Value: Descending Comments: Channel 8 will be used for the first outbound call, then channel 7 and so on.
ISPROXYUSED
Menu: Protocol Management\Protocol Definition\Proxy & Registration Display: Enable Proxy Recommended Value: Use Proxy Comments: A proxy must be used with Sphericall.
PROXYNAME
Menu: Protocol Management\Protocol Definition\Proxy & Registration Display: Proxy Name Recommended Value: Host of your primary MGC Comments: Due to problems with the way AudioCodes uses the results of DNS SRV lookups, Sphere recommends binding the MP-11X to a single MGC by specifying a hostname in this field. This recommendation may change with future releases of AudioCodes firmware.
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AUDIOCODES
Advanced Configuration
PROXYIP
Menu: Protocol Management\Protocol Definition\Proxy & Registration Display: Proxy IP Address Recommended Value: Host of your primary MGC Comments: Due to problems with the way AudioCodes uses the results of DNS SRV lookups, Sphere recommends binding the MP-11X to a single MGC by specifying a hostname in this field. This recommendation may change with future releases of AudioCodes firmware.
SIPGATEWAYNAME
Menu: Protocol Management\Protocol Definition\Proxy & Registration Display: Gateway Name Recommended Value: A value unique to each MP-11X, for example, MP-11X-1 Comments: A gateway name should be specified. If the gateway name is left unspecified, the MP-11X will use its IP address instead. If the IP address ever changes, example due to DHCP, the MP-11X will appear as a new gateway to Sphericall.
ISREGISTERNEEDED
Menu: Protocol Management\Protocol Definition\Proxy & Registration Display: Enable Registration Recommended Value: Enable Comments:
REGISTRARNAME
Menu: Protocol Management\Protocol Definition\Proxy & Registration Display: Registrar Name Recommended Value: Hostname of your primary MGC Comments: Due to problems with the way AudioCodes uses the results of DNS SRV lookups, Sphere recommends binding the MP-11X to a single MGC by specifying a hostname in this field. This recommendation may change with future releases of AudioCodes firmware.
REGISTRATIONTIME
Menu: Protocol Management\Protocol Definition\Proxy & Registration Display: Registration Time Recommended Value: 600 seconds Comments:
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Advanced Configuration
USERNAME
Menu: Protocol Management\Protocol Definition\Proxy & Registration Display: User Name Recommended Value: Must be the same as the gatway name
AUTHENTICATIONMODE
Menu: Protocol Management\Protocol Definition\Proxy & Registration Display: Authentication Mode Recommended Value: Per Gateway Comments: Setting the authentication mode to per gateway forces the MP-11X to register once for the entire gateway. More sophisticated configurations may require per endpoint.
CODERNAME
Menu: Protocol Management\Protocol Definition\Coders Display: Coder Name Recommended Value: g711Ulaw64k,20,0,$$,0 Comments: This value enables G.711 U law. Other CODECs, such as G.711 A law and G.729 may also be enabled.
ENABLECURRENTDISCONNECT
Menu: Protocol Management\Advanced Parameters\General Parameters Display: Enable Current Disconnect Recommended Value: Enable Comments: This setting causes the MP-11X to drop the call if loss of loop current is detected. This typically occurs when the party on the other side of the PSTN hangs up.
DISCONNECTONBROKENCONNECTION
Menu: Protocol Management\Advanced Parameters\General Parameters Display: Disconnect on Broken Connection Recommended Value: No Comments: If set to yes, the MP-11x will disconnect a call when it has not received any RTP packets for a configurable length of time. This causes premature disconnection of calls into Sphere voice mail and other devices that do not sent RTP packets when they are muted.
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AUDIOCODES
Advanced Configuration
CURRENTDISCONNECTDURATION
Menu: None (INI file parameter) Display: None (INI file parameter) Recommended Value: 300 milliseconds Comments: This setting sets the loss of loop current detection window to 200 to 600 milliseconds. This setting is not accessible via the web interface. It must be manually added to the INI file.
TIMETOSAMPLEANALOGLINEVOLTAGE
Menu: None (INI file parameter) Display: None (INI file parameter) Recommended Value: 100 milliseconds This setting forces the MP-11X to check the trunks for loss of loop current every 100 milliseconds. This settingis not accessible via the web interface. It must be manually added to the INI file.
ENABLECALLERID
Menu: Protocol Management\Advanced Parameters\Supplementary Services Display: Enable Caller ID Recommended Value: Enable Comments: Enables the delivery of Caller ID to the SIP Proxy in the Display name field of the INVITE From header.
PSTNPREFIX
Menu: Protocol Management\Routing Tables\IP to Hunt Group Routing Display: Dest Phone Prefix Recommended Value: *,1,*,xxx.xxx.xxx.xxx,1 Comments: This directs all outbound calls to the PSTN to the first hunt group.
TRUNKGROUP_1
Menu: Protocol Management\Endpoint Phone Numbers Display: 1 Recommended Value: 1-8, MP-11X, 1 Comments: This places all channels in the first hunt group.
ENABLECALLERID_<PORT>
Menu: Protocol Management\Endpoint Settings\Detect Caller ID from Tel
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Advanced Configuration
Display: Port <Port> Recommended Value: Enable Comments: This is a per channel setting. Enable it if your line delivers caller ID.
TARGETOFCHANNEL<PORT>
Menu: Protocol Management\Endpoint Settings\Automatic dialing Display: Port <Port> Recommended Value: <Auto Attendant Extension> Enable Comments: This is a per channel setting that determines where calls inbound from the PSTN on that channel are routed to.
ISTWOSTSTAGEDIAL
Menu: Protocol Management\FXO Settings Display: Dialing Mode Recommended Value: One Stage Comments: Use one stage dialing if the MP-11X is connected directly to the PSTN. Use two stage dialing if the MP-11X is connected to a PBX.
ISWAITFORDIALTONE
Menu: Protocol Management\FXO Settings Display: Waiting for Dial Tone Recommended Value: Yes Comments: Enabling this setting forces the MP11X to detect dialtone before dialing.
DTMFTRANSPORTTYPE
Menu: Advanced Configuration\Media Settingss\Voice Settings Display: DTMF Transport Type Recommended Value: Transparent DTMF Comments: This setting disables RFC2833 relay of DTMF digits and forces the digits to be carried in-band. This setting is required unless you have configured your Sphere system to use RFC2833.
MFTRANSPORTTYPE
Menu: Advanced Configuration\Media Settingss\Voice Settings Display: MF Transport Type Recommended Value: Transparent MF
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AUDIOCODES
Comments: This setting disables RFC2833 relay of DTMF digits and forces the digits to be carried in-band. This setting is required unless you have configured your Sphere system to use RFC2833.
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PLANNING
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Verify system requirements for AudioCodes MP104 and Sphericall. The Sphere system should be installed, configured and tested as fully functional.
OVERVIEW OF OPERATION
The AudioCodes MP104 Analog Media Gateway is a 4 port analog gateway that connects analog terminals, PBXs or key systems to the IP network using FXO connectivity. Using AudioCodes Analog Media Gateways, Sphere system Administrators can effectively deliver carrier-hosted converged services as well as enterprise-based applications. The AudioCodes MP104 SIP gateway is supported in Sphericall V5.1. Sphericall requires the AudioCodes version ID to be at or greater than 4.6. The following procedure outlines how to upgrade and configure an AudioCodes MP104 to version 4.60A.036.005 and allow it to check into a Sphere MGC.
Choose this method if the installer doesnt have access to the DHCP lease list.
1
Choose a computer that will be used for configuring the AudioCodes gateway. The computer will have its network disrupted during configuration so do not choose an inservice Sphericall Manager. Make sure the AudioCodes configuration files are accessible from the configuration computer (copied onto or accessible via the Sphericall CD). Power up the AudioCodes FXO MP104 gateway. A new AudioCodes FXO MP104 gateway uses a statically bound IP address of 10.1.10.11.
2 3
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AUDIOCODES
Planning
4 5 6
Change the configuration computers subnet mask to 255.255.0.0. Change its IP address to an address on the same subnet as the AudioCodes gateway (ex. 10.1.10.12). Either connect the gateway directly to a computer using a cross-over cable OR connect both the computer and gateway together using a switch / hub. Access the gateway Quick Setup screen using a web browser such as Internet Explorer. The URL is http://10.1.10.11.
Choose this method if the installer has access to the DHCP lease list.
1 2 3
Choose a PC for configuring the AudioCodes gateway that is in the same subnet as the gateway. Make sure the AudioCodes configuration files are accessible from the configuration computer (copied onto or accessible via the Sphericall CD). Power up the AudioCodes FXO MP104 gateway. When the AudioCodes FXO MP104 gateway is connected on a network that supports DHCP, it will be given an IP address within the DHCP address scope. Determine the gateways new IP address. The DHCP provided IP address can be determined by searching for the MAC address of the gateway within the IP address lease list of the DHCP server. The MAC address of the gateway is on a sticker located on the bottom side of the unit. Access the gateway Quick Setup screen using a web browser such as Internet Explorer. Use the IP address of the gateway as the URL (ex. http://172.16.15.35).
The gateways embedded web server will prompt for a username and password. Enter Admin for both (case sensitive). Download the software image. Go to Software Updates and browse to the AudioCodes file directory. Download 4.60A.036.005.cmp to the AudioCodes gateway. After 2 minutes, refresh the browser window. A new looking configuration pane will be displayed. From the Software Updates -> Load Auxiliary Files, load the following files:
a. INI file: SampleSIP.ini
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AUDIOCODES
Planning
IP CONFIGURATION
1 2
From the Quick Setup screen, enter either a valid static IP address & subnet mask that is appropriate for the target network. In the Default Gateway IP Address field, configure the IP address of the Sphericall Manager that is considered primary for this device. It may not necessarily be the primary Sphericall Manager for the system.
SIP PARAMETERS
3 4 5 6
Enter the name of the AudioCodes gateway in the Gateway Name field. When working with a Proxy server, set the Working with Proxy field to Yes. Enter the Proxy Name of the primary Proxy server in the Proxy IP Address field. When no Proxy is used, the internal routing table is used to route the calls. Enter the Proxy Name in the Proxy Name field. If Proxy name is used, it replaces the Proxy IP address in all SIP messages. This means that messages are still sent to the physical Proxy IP address but the SIP URI contains the Proxy name instead. Set Enable Registration to Enable. Disable = the MediaPack does not register to a Proxy server/ (default). Enable = the MediaPack registers to a Proxy server/Registrar at power up and every Registration Time seconds; The MediaPack sends a REGISTER request according to the Authentication Mode parameter.
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AUDIOCODES
Planning
Click the Reset button and click OK in the prompt; the MediaPack applies the changes and restarts.
Protocol Management. Protocol Definition. General Parameters tab. Change Fax Signaling Method to Fax Fallback. Advanced Configuration. Channel Settings. Fax/Modem/CID Settings tab. Change Fax Transport Mode to T.38 Relay.
Open the Automatic Dialing page: Protocol Management\Endpoint Settings submenu\Automatic Dialing.
In the Destination Phone Number field for a port, enter the telephone number to dial.
Incoming calls may be directed to a specific extension by specifying a valid extension. It is possible to specify the same extension for every port (e.g. the Auto Attendant) or a different extension for each port on this page (e.g. DID lines).
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AUDIOCODES
Advanced Configuration
It is possible to use the default routing specified in the Sphericall Administrator application by specifying an invalid extension for one or more of the ports on the Automatic Dialing page. For example, if the nonexistent extension "xyz" is specified on the Automatic Dialing page and the domain is "spherecom.com", when an incoming call is received, the MP11x will send an INVITE to "xyz@spherecom.com". Since "xyz" does not exist on the system, the MGC will apply the default route to the call.
3
Enable [1] When a port is selected, when making a call, the number in the Destination Phone Number field is automatically dialed if phone is off hook (for FXS gateways) or ring signal is applied to port (FXO gateways). Disable [0] The automatic dialing option on the specific port is disabled (the number in the Destination Phone Number field is ignored). Hotline [2] When a phone is offhook and no digit is pressed for HotLineDialToneDuration, the number in the Destination Phone Number field is automatically dialed (applies to FXS and FXO gateways).
4
Repeat step 3 for each port you want to use for Automatic Dialing. Click the Submit button to save your changes.
ADVANCED CONFIGURATION
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While your settings may vary, Sphere Communications has changed the following Advanced MP104 settings for integration with the Sphere system.
CHANNELSELECTMODE*
Menu: Protocol Management\Protocol Definitions\General Parameters Display: Channel Select Mode Recommended Value: Descending Comments: Channel 8 will be used for the first outbound call, then channel 7 and so on.
ISPROXYUSED
Menu: Protocol Management\Protocol Definition\Proxy & Registration Display: Enable Proxy Recommended Value: Use Proxy Comments: A proxy must be used with Sphericall.
PROXYNAME
Menu: Protocol Management\Protocol Definition\Proxy & Registration
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AUDIOCODES
Advanced Configuration
Display: Proxy Name Recommended Value: Host of your primary MGC Comments: Due to problems with the way AudioCodes uses the results of DNS SRV lookups, Sphere recommends binding the MP104 to a single MGC by specifying a hostname in this field. This recommendation may change with future releases of AudioCodes firmware.
PROXYIP
Menu: Protocol Management\Protocol Definition\Proxy & Registration Display: Proxy IP Address Recommended Value: Host of your primary MGC Comments: Due to problems with the way AudioCodes uses the results of DNS SRV lookups, Sphere recommends binding the MP104 to a single MGC by specifying a hostname in this field. This recommendation may change with future releases of AudioCodes firmware.
SIPGATEWAYNAME
Menu: Protocol Management\Protocol Definition\Proxy & Registration Display: Gateway Name Recommended Value: A value unique to each MP104, for example, MP104-1 Comments: A gateway name should be specified. If the gateway name is left unspecified, the MP104 will use its IP address instead. If the IP address ever changes, example due to DHCP, the MP104 will appear as a new gateway to Sphericall.
ISREGISTERNEEDED
Menu: Protocol Management\Protocol Definition\Proxy & Registration Display: Enable Registration Recommended Value: Enable Comments:
REGISTRARNAME
Menu: Protocol Management\Protocol Definition\Proxy & Registration Display: Registrar Name Recommended Value: Hostname of your primary MGC Comments: Due to problems with the way AudioCodes uses the results of DNS SRV lookups, Sphere recommends binding the MP104 to a single MGC by specifying a hostname in this field. This recommendation may change with future releases of AudioCodes firmware.
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Advanced Configuration
REGISTRATIONTIME
Menu: Protocol Management\Protocol Definition\Proxy & Registration Display: Registration Time Recommended Value: 600 seconds Comments:
USERNAME
Menu: Protocol Management\Protocol Definition\Proxy & Registration Display: User Name Recommended Value: Must be the same as the gateway name Comments:
AUTHENTICATIONMODE
Menu: Protocol Management\Protocol Definition\Proxy & Registration Display: Authentication Mode Recommended Value: Per Gateway Comments: Setting the authentication mode to per gateway forces the MP104 to register once for the entire gateway. More sophisticated configurations may require per endpoint.
CODERNAME
Menu: Protocol Management\Protocol Display: Coder Name Recommended Value: g711Ulaw64k,20,0,$$,0 Comments: This value enables G.711 U law. Other CODECs, such as G.711 A law and G.729 may also be enabled.
ENABLECURRENTDISCONNECT
Menu: Protocol Management\Advanced Parameters\General Parameters Display: Enable Current Disconnect Recommended Value: Enable Comments: This setting causes the MP11X to drop the call if loss of loop current is detected. This typically occurs when the party on the other side of the PSTN hangs up.
CURRENTDISCONNECTDURATION
Menu: None (INI file parameter)
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Advanced Configuration
Display: None (INI file parameter) Recommended Value: 300 milliseconds Comments: This setting sets the loss of loop current detection window to 200 to 600 milliseconds. This setting is not accessible via the web interface. It must be manually added to the INI file.
TIMETOSAMPLEANALOGLINEVOLTAGE
Menu: None (INI file parameter) Display: None (INI file parameter) Recommended Value: 100 milliseconds This setting forces the MP104 to check the trunks for loss of loop current every 100 milliseconds. This setting is not accessible via the web interface. It must be manually added to the INI file.
ENABLECALLERID
Menu: Protocol Management\Advanced Parameters\Supplementary Services Display: Enable Caller ID Recommended Value: Enable Comments: Enables the delivery of Caller ID to the SIP Proxy in the Display name field of the INVITE From header.
PSTNPREFIX
Menu: Protocol Management\Routing Tables\IP to Hunt Group Routing Display: Dest Phone Prefix Recommended Value: *,1,*,xxx.xxx.xxx.xxx,1 Comments: This directs all outbound calls to the PSTN to the first hunt group.
TRUNKGROUP_1
Menu: Protocol Management\Endpoint Phone Numbers Display: 1 Recommended Value: 1-8, MP104, 1 Comments: This places all channels in the first hunt group.
ENABLECALLERID_<PORT>
Menu: Protocol Management\Endpoint Settings\Detect Caller ID from Tel Display: Port <Port> Recommended Value: Enable
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Advanced Configuration
Comments: This is a per channel setting. Enable it if your line delivers caller ID.
TARGETOFCHANNEL<PORT>
Menu: Protocol Management\Endpoint Settings\Automatic dialing Display: Port <Port> Recommended Value: <Auto Attendant Extension> Enable Comments: This is a per channel setting that determines where calls inbound from the PSTN on that channel are routed to.
ISTWOSTSTAGEDIAL
Menu: Protocol Management\FXO Settings Display: Dialing Mode Recommended Value: One Stage Comments: Use one stage dialing if the MP104 is connected directly to the PSTN. Use two stage dialing if the MP11X is connected to a PBX.
ISWAITFORDIALTONE
Menu: Protocol Management\FXO Settings Display: Waiting for Dial Tone Recommended Value: Yes Comments: Enabling this setting forces the MP104 to detect dialtone before dialing.
DTMFTRANSPORTTYPE
Menu: Advanced Configuration\Media Settings\Voice Settings Display: DTMF Transport Type Recommended Value: Transparent DTMF Comments: This setting disables RFC2833 relay of DTMF digits and forces the digits to be carried in-band. This setting is required unless you have configured your Sphere system to use RFC2833.
MFTRANSPORTTYPE
Menu: Advanced Configuration\Media Settings\Voice Settings Display: MF Transport Type Recommended Value: Transparent MF Comments: This setting disables RFC2833 relay of DTMF digits and forces the digits to be carried in-band. This setting is required unless you have configured your Sphere system to use RFC2833.
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Advanced Configuration
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Advanced Configuration
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The Sphere system should be installed, configured and tested as fully functional. Refer to the (1)Spectralink Facility Preparation and (2)Installation and Operation Manuals for installation planning, setup, package contents, safety and conditions of use.
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REQUIRED MATERIALS
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Cross Connect Block - required to connect the telephone switch ports and the base stations to the MCU. 25 Pair Cables - RJ-21 male at MCU end, required to connect the MCU to the cross-connect blocks.
SPECTRALINK OVERVIEW
The Link Wireless Telephone is a durable and feature-rich handset for workplace applications. The Link Wireless Telephone integrates with the Sphere system to provide advanced calling features throughout the workplace.
Note: To Sphere-related hardware, the Spectralink phone takes on the
characteristics of an analog phone. However, the Spectralink phone adheres to the definitions of voice-over-wireless technology and is subject to the requirements of said technology. As previously stated, refer to Spectralinks documentation for all operational issues.
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Required Materials
VG3 PhoneHub
24 23 22 21 20 19 18 17 16 15 14 13 12 11 10 9 8 7 6 5 4 3 2 1 25
VG3-PB2430
FA ILO VER
MODE
SELECT
10/100 ETHERNET
STATION LINES
Base Stations
LAN
Switch
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Required Materials
Spectralink
PWR OUT IPC IN L I N K O K A C T C O L Network E R R O R 1 2 3 4 5 CONN A
STATUS
RS-232
Table 15.1
Definition
Not Used in integration with Sphere system Not Used in integration with Sphere system Connect Spectralink MCU to organizations network Flashes when the system has detected an error. When flashing, check the Status LEDs for an error code. Indicate system error messages and status. Connects to the AC adapter to supply power to the system. RJ-21 connector to the cross-connect demarc block. 9-Port Serial Connector used to interface with Spectralink MCU for device configuration.
BASE STATIONS
Base Stations act as radio transceivers to provide the communications signal between the wireless telephone and the MCU. Base Stations are slightly larger than a smoke detector and are typically mounted on the ceiling, in strategic locations throughout the facility. A single Base Station can provide radio coverage for an area of 5,000 to 50,000 square feet depending on building obstructions. Base Stations may be located up to 2,200 cable feet from the MCU. When a Wireless Telephone user makes or receives a call, the Wireless Telephone and Base Station establish a digital radio communication link. As the user moves around the coverage area, calls are handed off to the Base Station that is able to provide the best radio signal (typically the closest Base Station). These handoffs involve the Wireless Telephone
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Required Materials
establishing a communication link with another Base Station and dropping the previous link.
Note: Spectralink provides a list of access points that are compatible with the
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16
The Sphericall Phone type library exposes Sphericall objects that enable the programmer to create their own Graphical User Interface(s) for their telephones. The Sphericall Phone Type Library has built-in Help files on how to use the Sphericall Phone Objects, but it does not teach programming skills. Each Help file shows an example, gives properties, methods and events, and indicates Applies To. This online Help file will be your main resource for using the Sphericall Phone Type Library (the help file name is: Phonetlb.hlp; the contents file name is: Phonetlb.cnt). Consult your programming manuals and on-line Help for programming ideas. If you wish to integrate this Help with the Visual Basic Help, you must add the following line to the Visual Basic contents file (i.e. VB.cnt) after the final :Index statement. :include phonetlb.cnt Use the SCPhone Object to create an instance of the Sphericall Phone object, dial numbers, pick up extensions, retrieve an individual call object, respond to phone events, and access line and title properties. Use the SCCalls Object to step through all active calls on the SCPhone Object. Use the SCCall Object to retrieve information about a single call on the SCPhone Object. This object also handles answering, transferring, hanging up, placing a call on hold, playing sounds, and dialing additional digits after the call has been identified.
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CONSTANTS
CALLSTATECONSTANTS
The CallStateConstants are enumerated values representing the status of a call. They are typically used with the State and PreviousState properties of SCCall. When the State property of the current call matches a CallStateConstants constant, the StateName of the call matches the indicated description.
Constant
callStateUnknown callStateRinging callStateDialTone callStateDialing callStateProceeding callStateRingback callStateBusy callStateSpecialInfo
Value
0 1 2 3 4 5 6 7
Description
Unknown Ringing Dial Tone Dialing Proceeding Ringback Busy Special Info
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Getting Started
8 9 10 11 12 13
Be sure to take advantage of the example code fragments within the Help files provided by Sphere. Before long, you will create your own integration of Sphericall with your most commonly performed tasks within the application of your work. Microsoft publishes a printed VBA reference manual, but it doesn't come with most VBA applications. The manual is included with the Developer's Edition of Office.
GETTING STARTED
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The Sphericall COM type library provides an API where developers can create custom applications utilizing call information and control. Using COM allows the developer to choose the programming language such as Visual Basic, VB script, javascript , or C++.
Login into the PC with a user account that has local administrative privileges . Install the Sphericall Desktop. Using the Sphericall Administrator, configure user rights so that the user account has full privileges to open one or more stations. Start the Sphericall Desktop and open a station line. Start an Internet Explorer. Go to a web site that provides a phone number or use a name lookup web site such as people.yahoo.com or www.switchboard.com. Highlight a number, right-mouse-click and Select Sphericall Dial.
The Sphericall Desktop will receive the number via the COM interface and then dial it. When the Sphericall Desktop is started, it will load and register the phone.tlb type library. Registration requires write access to HKEY_CLASSES_ROOT registry key, which in turn requires the user to have administrative access to the PC.
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Getting Started
In a future version of Sphericall (timeline is not yet available), you can anticipate a features such as a status message indicating the result of loading and registering the type library will be printed in the Sphericall Desktop log file. A status message will also be added to the log files for every command invoked using the COM API. Desktop log files are stored in C:\Documents and Settings\Logged_In_Username\Application Data\Sphere\Desktop\Logs. If the Sphericall Desktop is not running when a COM call control command is invoked, the Sphericall Desktop will be started. It is possible for applications written using the COM interface to block thereby keeping control of the Sphericall Desktop. The Sphericall Desktop will release control if the application is blocked for more than 5000 milliseconds. This value can be changed through [HKEY_CURRENT_USER\Software\Sphere\Phone\COMInterface] "MessagePendingDelay" The sample applications (Access, Visual Basic, Web, DesktopCOMTest ) are located in \Server\Data\samples\Desktop COM API\install. Visual Basic applications require the VB runtime. Installing the sample applications will install the VB runtime. The MS Access sample requires MS Access to run. The descriptions of the Sphericall Objects and their methods is located in C:\Program Files\Sphere\phonetlb.hlp.
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Getting Started
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SIP TRUNKING
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17
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OVERVIEW OF SIP
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Session Initiation Protocol (SIP) is a text-based protocol, similar to HTTP and SMTP, for initiating interactive communication sessions between users. Such sessions include voice, video, chat, interactive games, and virtual reality. The Sphericall Manager is designed to optimize the use of Session Initiation Protocol for communications via SIP station devices as well as SIP trunk facilities, or soft trunks. Created in 1996 by IETF (Internet Engineering Task Forcethe same organization that created TCP/IP and HTTP), SIP was originally intended to enable the establishment of media (audio or video) sessions between users. Since that time, it has evolved to cover a wide range of real-time collaboration functionalities. In the simplest of terms, SIP provides a mechanism for setting up generic sessions of information exchanged among disparate endpoints across the IP network. Within the Sphericall Manager, SIP allows external systems to participate in calls with the system. Early on, the Sphericall Manager interacted with the Windows Messenger client using the SIP standard protocol to connect their application and signaling needs. SIP has achieved wide adoption throughout the telecommunications and enterprise markets for its ability to streamline communication session control and provide cross-application interoperability. Although there are other protocols designed to have similar functionality, SIP is said to be simpler and consumes fewer resources, the protocol of choice for truly converged networks. Further, SIP can be harnessed to do various tasks that are currently accomplished by multiple protocols, many of which are proprietary. Voice over IP (VoIP) networks use SIP as a call control functionality to connect phones so the phones can exchange media information through a media exchange protocol, like RTP. In this context, "connect" includes resolving the address of the destination and negotiating the types of media to be exchanged. For address resolution, SIP provides functionality similar to what Domain Name System (DNS) provides for URLs. DNS maps human-friendly, fully qualified domain names, like www.spherecom.com, to numeric IP addresses required for communication over an IP network. Similarly, SIP maps a URL or a phone number (xxx-xxx-xxxx in the United States) to an IP address to which the phone can send media. SIP also provides additional functionality that DNS does not provide. For example, people can have multiple IP phones that they use in different locations. A SIP server can dynamically change the IP address it returns for a specific URL or phone number depending on whether the recipient is in the office, car, home or other location. Because the IP address changes, calls are routed to the phone that the person is using. In addition to resolving addresses of destination phones, SIP uses Session Description Protocol (SDP) to negotiate media formats that both phones support. The content of the SIP invitation message that a phone generates during call setup
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contains an SDP message that defines the media formats that the source phone supports. The destination phone examines the SDP message and creates a SIP response that says which media formats the call should use. In terms of communication between SIP endpoints and the structure of SIP packets, SIP resembles HTTP in that it is a request-response protocol. However, unlike HTTP, SIP allows for additional responses to a single request. In response to an invitation, a phone can initially return a "ringing" response, and then an "ok" response when the phone is picked up. VoIP networks that use SIP generally provide a few common services that phones use to carry out SIP conversations. These services provide much of the additional functionality for which SIP was designed, like robust address resolution to multiple locations for a single URL or phone number. These services refer to server functionality, not to where or how the functionality is implemented. An example: in many cases the same physical server provides all of these services. The following additional services are commonly provided by SIP: Registrar and Location. One of the first SIP operations an IP phone performs is to register with a registrar server, by providing a URL or phone number and a corresponding IP address. The registrar server stores this information and often provides it to proxy and redirect servers to resolve URLs and phone numbers to IP addresses. Proxy and Redirect. Proxy and redirect servers assist IP phones in routing SIP messages to a destination. The mechanisms they use to accomplish this goal are slightly different. A proxy server accepts SIP requests and forwards them to another SIP device, which can be an endpoint like a phone, or another server (which can then forward to a phone or another server, and so on). As far as the original client is concerned, the proxy server handles everything required to get the request to the destination. In contrast, a redirect server does not forward requests. Instead, it returns an IP address to the IP phone; the IP phone then uses the returned address to submit the request to the correct location (which can be a proxy server or the actual phone). For More Information: Additional reading of the following documents may help you to understand further the SIP environment and goals: Documents, wiki and white papers at: www.sipforum.org or sip.edu Cookbook http://mit.edu/sip/sip.edu/index.shtml
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The Sphericall system needs to identify the terminal that inbound calls are being placed from in order to apply policy such as inbound routing, capacity management, etc. This following is the logic the Sphericall Manager uses to locate a SIP terminal: Definition: SIP URI = userinfo@hostname:port
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FROM Header
trunks since the FROM field contains the caller ID of the incoming call.
2
Contact Header
hostname:port is compared to the Outbound Proxy if configured, otherwise the Service Provider Domain of the Service provider information. If the hostname:port is an ip address it is compared exactly to what is configured. If the hostname:port is not an IP address, a partial compare is performed against the Service Provider information. For example, the hostname "horatio.Nlab.spherecom.com" would match the Service Provider information "Nlab.spherecom.com".
Note: In both cases the port must match.
3
To Header
Userinfo is compared to the DID maps configured for SIP trunks. First match is found is used if there is overlapping DID configuration.
4
Authorization Header
The credentials included in the Authorization Header are compared against the credentials configured in the Authorization window for the trunk. The Sphericall Manager will challenge the sender to obtain credentials via the Authorization Header if no match is found using the above tests.
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The following three configuration scenarios define a number of the installation scenarios that might be seen in the initial stages of NEC Sphere SIP trunking implementation. SCENARIO 1: Sphericall to SIP carrier - softtrunking This is a pure SIP softtrunking to an outside service provider. This implementation will routinely use Outbound registration type. SCENARIO 2: Sphericall to Sphericall - softtrunking tie line SIP trunking as a tie line between existing Sphere systems. This implementation will routinely use None registration type. SCENARIO 3: Sphericall to Centurion Call Center - softtrunking with tie line to call center SIP trunk as a tie line to a call center. This implementation will routinely use Outbound registration type.
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2 3 4 5 6 7 8
Review the User Agents listed in the window. If the SIP endpoint you are using is not listed in this window, you must add it. Click Add. Enter the User Agent name. Select Endpoint Type: REQUIRED. The Sphere system requires this information in order to know how to treat it or what features to apply per endpoint. Enter an Agent Description that is appropriate for the Name & Endpoint Type. Click Apply. Note: 1) Those User Agents listed in the SIP dialog window that also have the
Default checked, are those User Agents created into the system by default. These will remain in the system. User Agents added by system administrators are not indicated with a check in the Default column.
Note: 2) If the name of a non-default user agent matches the name of a default user
agent, the Agent Name, Agent Description and Endpoint Type fields cannot be edited. Conversely, if multiple user agent entires exist that have the same name, but none are marked as default, changing any field (other than Version) will change the same corresponding field in the other same-named User Agent entries. The following fields are customizable for entering a non-default user agent:
Table 17.1 User Agent Profile Descriptions
Possible Values
Supported Unsupported (default)
Description
The MGC sends the NOTIFY request to answer/hold/unhold a call remotely (Sphericall Desktop or Web Services).
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Possible Values
Allowed Disallowed (default)
Description
MGC sends a NOTIFY request with terminated (reason=timeout) subscription-state when a SUBSCRIBE request with non-empty to-tag is received and the corresponding subscription is not found. MGC sends a special value of the parameter (Auto Answer, answer-after etc) in the outgoing INVITE request to inform the SIP station to answer immediately. Call Manager creates a SIP phone when not found in the database. Since in 5.2.1+ all Polycom SIP phones are precreated by the system administrator (just like the SIP soft trunks), this capability ensures that these phones are not created by the MGC when not found in the database. When a Polycom phone sends its first REGISTER to check into an MGC, the MGC immediately sends a "503 Service Unavailable" response with a Retry-After timeout of 300 seconds. In the mean time MGC obtains phone configuration from the database. If the configuration from the database is not available, MGC continues sending "503 Service Unavailable" message in response to the REGISTER requests. MGC uses the appropriate method to find the terminal associated with the incoming INVITE request from the endpoint.
Click-To-Dial
Ring Callers Phone First (default) Use answer-after ??param (INVITE:: Call-Info Header) Use auto-answer value (INVITE:: Call-Info Header)
Endpoint Created By
Authentication Info Default (default) DID Mapping From Header URI Outbound Contact URI P-Asserted-Identity Header URI (currently supported)
Hardware Address
SIP has the capability to search for a hardware address in the REGISTER request. (Quintum products only)
MGC sends an unsolicited NOTIFY request when the MWI state changes.
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Possible Values
Supported (default) Unsupported
Description
When the value is set to Supported, the MGC will not send a NOTIFY request to the endpoint even if there is a change in the MWI status. If the value is set to Unsupported MGC will send the unsolicited NOTIFY (out of dialog) request when there is a change in the MWI status even if the endpoint has not sent a SUBSCRIBE request. Setting specifies the maximum packet size Sphericall Media Server should send to the far-end. Setting specifies which SIP endpoints support or do not support OPTIONS request. MGC does not send REFER to initiate a click-to-dial call. MGC sends REFER to initiate a transfer. MGC sends the NOTIFY request to reboot a station. MGC sends the QHeader (Question header in a URI) in the Refer-To header in the REFER request. MGC sends Video SDP in the outgoing INVITE.
OPTIONS Request
Supported Unsupported (default) Supported (default) Unsupported Supported Unsupported (default) Supported (default) Unsupported
Remote Reboot
Video
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These tables essentially provide the functionality provided by the MGSetting table, but at a much less overhead. These tables also provide the following benefits: When a Sphericall system is upgraded (pre-6.0 to 6.0+), Sphericall automatically creates UserAgentVersionHubAssociation for all Outbound/None registration type devices. It also creates a binding in UserAgentVersionHubAssociation table for those SIP devices (irrespective of registration type) that have an entry in the deprecated UserAgentHubAssociation table. This ensures that the SIP devices keep functioning after the upgrade. When a Sphericall system is upgraded (6.0 to 6.0+), Sphericall will not change the existing UserAgentVersionHubAssociation bindings. Therefore, the SIP devices overriding the default behavior will continue to work the same way after the
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upgrade. However, Sphericall may change the default values of the Parameters configured in the Parameter table (this is similar to that of the MGSetting defaults which may infrequently get changed on an upgrade). When the firmware of a SIP device is upgraded, if the SIP device has a forced binding in the UserAgentVersionHubAssociation table, the DbServer/MGC do not update the binding for the new firmware and the SIP device continue the work same way in the MGC. For Outbound/None registration type endpoints DbServer and MGC never update the UserAgentVersionHubAssociation binding even if the SIP device is reporting a completely different User-Agent/Firmware-Version than what is configured in the UserAgentVersionHubAssociation table. A new User-Agent/Firmware-Version can be created and assigned to a custom ParameterProfile and then this User-Agent/Firmware-Version can be bound to the UserAgentVersionHubAssociation and set to "forced" type binding. This feature provides a similar level of control provided by the MGSetting table. SIP station upgradesUpgrades to User Agents with firmware are automatic and apply to all the endpoints on the system. SIP trunk upgradesUpgrades to User Agents are not automatic.
From the Sphericall Administrator application: Open the General System properties. Select the SIP tab. Scroll to view the User Agents. Click the far right column of the User Agent to be removed. Click Remove. Repeat for other User Agents. Click OK to exit.
Configuration Needed
Must create trunk on both ends of the softtrunk. These trunks will have the Service Provider tab in the software.
Uses
Most SIP softtrunks are generally created in this way.
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Registration Type
Outbound Registration
Configuration Needed
Must create trunk via configuration and softtrunk will be auto-created on the other send. These trunks will have the Service Provider tab in the software. Do nothing with configuration. This softtrunk will just appear in the Sphericall Administrator Trunks section. The far side will have information to send the registration. These trunks will not have the Service Provider tab.
Uses
Tie-lines may be created in this way. Other, non-standard, softtrunks may be created with this method.
Inbound Registration
Tie-lines may be created in this way. Call center softtrunks are often created with this method.
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The above scenario of a SIP solution has the following characteristics: This company has one or more individual connection(s) to the SIP Service Provider (SSP); rather than contracts with an ISP, the local phone services, long distance services, etc. are secured through the SIP service provider. Further connections beyond SSP to the PSTN (Public Switched Telephone Network) are the responsibility of the SSP. Billing and services are simplified.
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Gateway CPE (ALG) (Customer Premise Equipment: Application Layer Gateway or VoIP Aware Gateway) is placed on the premise by SSP and typically maintained by SSP. Features and services available through SSP vary by provider.
YOU MUST FIRST VERIFY OR ADD THIS SIP ENDPOINT AS A USER AGENT TO THE SYSTEM. Refer to the beginning of this chapter for this procedure for all SIP endpoints. Refer to Book 1: Planning & Preparing the Sphere System for information on ordering and planning SIP services. Once you have your SIP services delivered and the Customer Premise Equipment (CPE) is onsite (typically an ALGApplication Layer Gateway), connect the CPE to the local area network (straight-through RJ-45 cable from the CPE to the Ethernet network switch).
Application Layer Gateways (NAT traversal) An ALG can allow firewall traversal with SIP. If the firewall has its SIP traffic terminated on an ALG then the responsibility for permitting SIP sessions passes to the ALG instead of the firewall. An ALG can solve another major SIP headache: NAT (network address translation) traversal. Basically a NAT with built-in ALG can re-write information within the SIP messages and can hold address-bindings until the session terminates. This allows for data inside the SIP packet to be rewritten with a public IP address to allow the far end to route calls back to the original party.
Note: ALG hardware: some SIP trunk service providers will provide this piece of
hardware as part of their service (like a smartjack). Other providers leave this piece of equipment up to the end customer to purchase. Another piece of equipment that can also be used in this scenario is called a Session Border Controller (SBC). Sphere has worked with the following ALGs in the field: Ingate, Adtran and ATI.
4
During the ordering process of your SIP trunk service, you must secure the following information:
Account Number Service Provider Domain name Outbound Proxy Registration Type Contact Domain Primary MGC (Sphericall Manager) Secondary MGC (Sphericall Manager) Password Realm
5
In some cases, the service provider may only make changes or additions to your services or the service order through one authorized individual.
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RFC2833 MUST BE OFF on the Media Streams tab of General/System Properties/Media Streams. ONLY G711 is enabled on General/System Properties/Media Streams. On Media Server Properties, be sure that the Codec Settings are DISABLED from the system default (be sure to de-select this checkbox).
Select the Trunks tab. Right-click and select Add. Choose Add Softtrunk.
Description: Any printable ASCII character except double quote or backslash. This field is a description field that specifies the display-name in the From and To headers of Outbound REGISTER requests. Often it will be the phone number that is assigned to this service. Example: 8477939942 Account: Provided to you by SIP carrier (i.e. could also be phone number that is assigned to this service). Example: 8477939942 The Account field specifies the userinfo portion of the SIP URI in the From and To headers of outbound REGISTER requests. The Account field may contain any printable character except
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the following: space, pound sign #, percent sign %, colon :, less than <, greater than >, at sign @, square brackets [], backslash \, caret ^, brace brackets {}, pipe | and grave accent `.
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Service Provider Domain: Provided by the SIP service provider It is the destination identification information (its who you are sending TO). Example: 777445.sip.company.net. The Service Provider Domain field (SPD) may be filled with an IP address, DNS host name or DNS domain name. The SPD specifies the hostname portion of the SIP URI in the From and To headers of outbound REGISTER requests. The SPD also specifies the destination for outbound SIP requests if the Outbound Proxy (OP) is blank. A DNS lookup is performed on the contents of the SPD if it does not contain an IP address. The type of DNS lookup performed is specified by the DnsNaptrServiceEnabled and DnsSrvServiceEnabled settings (see Note below). Outbound Proxy: Provided by the SIP service provider Example: sipstack-5.mnl2.sipcarrier.net The Outbound Proxy field (OP) may be filled with an IP address, DNS host name, DNS domain name or it may be left empty. If the OP is not empty, all outbound SIP requests are sent to the specified destination. A DNS lookup is performed on the contents of the OP if it does not contain an IP address. The type of DNS lookup performed is specified by the DnsNaptrServiceEnabled and DnsSrvServiceEnabled settings (see Note below). If the OP is empty, the destination for outbound SIP requests is determined by the Service Provider Domain field.
Summary of SPD and OP The SPD and OP determine where the Sphericall system sends outbound SIP requests. The OP field would typically be used when the administrator wants to force all outbound SIP messages through a specific device, for example, a SIP Session Border Controller (SBC).
Note: The following two System Initialization Settings affect settings for the SPD and
OP:
DnsNaptrServiceEnabled The DnsNaptrServiceEnabled MGC setting specifies whether the MGC will perform DNS NAPTR queries on the SPD/OP. This setting defaults to true. The administrator may optimize MGC performance by changing this setting to false if the SPD/OP specifies a host name or if the authoritative DNS server for the SPD does not support NAPTR queries. DnsSrvServiceEnabled The DnsSrvServiceEnabled MGC setting specifies whether the MGC will perform DNS SRV queries on the SPD/OP. This setting defaults to true. The administrator must change this setting to false: if the SPD/OP specifies a host name, if the SPD is not a valid DNS domain name with an SRV record, or if the authoritative DNS server for the SPD does not support SRV queries.
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Port information: use the default (5060) unless otherwise specified by the SIP service provider. Registration type: Recommended for softtrunk to service provider: OUTBOUND The registration type is governed by the service provider.
Inbound - the SIP trunk sends a register request to the Sphericall system. When the SIP trunk first initiates contact by sending a REGISTER request, the MGC automatically creates a hub, trunk and outside service. The Service
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Provider tab does not appear in the Hub device properties dialog box since the SIP trunk provides all the required information in the REGISTER request. Outbound - the Sphericall system sends register request to the SIP trunk; all fields on the Service Provider tab (trunk device properties) are used. None - issues no registration; neither the SIP trunk nor the Sphericall system send a register request; all fields on the Service Provider tab except the Contact Domain are used (trunk device properties). Some SIP trunk service providers require the Sphere trunk to register with their system. This registration can be set to None, Outbound or Inbound depending on the requirement of the SIP trunk Service provider.
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Primary MGC:
Secondary MGC: The Primary and Secondary MGC fields specify which MGCs are primary and secondary for the SIP trunk. The Secondary MGC assumes control of the trunk should the Primary fail. If the CD specifies an IP address or DNS host name, the Secondary MGC field should be filled with "None." In this case, specifying a Secondary MGC has limited value since the external system will not be able to locate the Secondary MGC to send SIP requests to. In practice, calls inbound to the Sphericall system will fail. Calls outbound from the Sphericall system may fail. If the CD specifies a DNS domain name, the DNS SRV records must match the contents of the Primary and Secondary MGC fields. Specifically, the "first" SRV record must point to the Primary MGC and the "second" SRV record must point to the Secondary MGC. The order in which the SRV records are returned are determined by the priority and weight values assigned to the SRV records in DNS. The external system must attempt to contact the MGC pointed to by the SRV record with the lowest priority first. If multiple SRV records have the same priority, the external system must attempt to contact the MGC pointed to by the SRV record with the highest weight first. Since Microsoft's DNS server returns different orderings for SRV records with the same priority from query to query, we recommend setting the (priority, weight) of the SRV record for the Primary and Secondary MGC to (0, 0) and (1000, 0) respectively. If the CD specifies a DNS domain name, that name must be resolvable from the external system. For example, we cannot use internal_domain.YOURDOMAIN.com as the CD for a SIP Service Provider or carrier since internal_domain.YOURDOMAIN.com cannot be resolved anywhere other than the internal Sphere network.
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Contact Domain: You provide this information It is the identification information YOU publish to the far end (you are identifying who you are). Example: 11.2.27.101 The Contact Domain field (CD) specifies the hostname portion of the SIP URI in the Contact header of outbound REGISTER requests. In other words, the CD determines how the external system locates an MGC to send SIP requests to. The CD may be filled with an IP address, DNS host name or DNS domain name. If filled with an IP address or host name, the external system will be bound to a single MGC with no redundancy. The CD should only be filled with a DNS domain name if the external system supports the DNS NAPTR and SRV lookup procedures specified by RFC 3263. Notice that the hostname portion of the SIP URI in the Contact header of all outbound non REGISTER requests is filled with the IP address of the MGC sending the request. This binds the SIP dialog to the MGC.
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Determines the preferred network transport protocol for SIP signaling (TCP, TLS, UDP (default)). Note: For SIP trunks that use registration types Outbound or None, this setting specifies what transport is used to send SIP requests such as REGISTER and INVITE to the SIP Trunk. For SIP stations and trunks that use registration type Inbound, the MGC sends SIP requests via the transport the SIP endpoint specifies in its REGISTER request.
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Sphere has included for selection most of the common types of User Agents. If the User Agent your system is using is not offered here, you must create it in the System General Properties (SIP) window, then apply it to the User Agent associated with the trunk. The following vendors specifically should use the Generic SIP Trunk User Agent:
Bandtel Clear SIP Constant Touch Global Crossing Netlogics Net Vortex Starvox Xtra (Spain)
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DNS Test:
DNS test will query DNS resolving NAPTR records. The DNS "Service Provider" and "Outbound Proxy Addresses will always be checked for A records and SRV records (records along with priorities and weights), as well as Contact Domain records. It will be up to the administrator to interpret the results and what affect, if any, it has on the SIP trunk. The DNS NAPTR query is not attempted when the Sphericall Administrator application is run under Vista.
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Click OK.
The creation of the softtrunk is complete. SIP Trunk MAC Address Assignment When a trunk checks into a Sphericall system for the first time, the MGC automatically creates the necessary database records for it. As part of the record creation process, the MGC assigns a unique name to the trunk. For SIP trunks, the MAC address assigned to the trunk depends upon the registration type. For outbound registrations and registrations of type none, the MGC uses "Account@Service_Provider_Domain." For inbound registrations, the MGC uses the SIP URI of the REGISTER request To header. It is important to note that if the URI changes, the MGC will create new records for the trunk the first time it sends a REGISTER with the new URI.
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Once creation of the SIP softtrunk is complete, the following information will be available on the Hub Properties:
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Name: Name of device (can be renamed for user-friendliness). MAC Address: Assigned to the device by the system. Stations/Trunks: Information on number of stations or trunks. LAN: Hub belongs to LAN displayed in this field. Localization Setting: Country or local language specification as defined in the Localization Setting profiles. Description: Information in this field can be edited. Hub Number: Automatically numbered by the system (recommended that this number remains unchanged). Last Check In Time: Information is updated upon each check-in. Firmware Version: None. Device Type: Default device type for Softtrunk is Unknown SIP Agent.
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Verify the information that appears on this window. It should be the same as the initial Add SoftTrunk dialog entries.
This information will vary depending on the service provider, their design and use of the SIP information. You may edit this information if necessary (example: change the Description, Port and Registration Type or other editable fields).
Note: The "Account" and "Service Provider Domain" were left read-only because
Select the Trunks tab from the main Sphericall Administrator application. Expand the tree of trunk devices listed in this window. Expand the + sign by the trunk device you wish to open. Double-click to open the Trunk Properties.
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Name: Type name of softtrunk Hardware ID: Usually assigned dynamically by the system. Telephony Area: Assign or change Telephony area here. Zone: Assign or change Zone here. Total Capacity: Configure based on services provided by SIP provider. Inbound Capacity: Determine based on services provided, see below. Outbound Capacity: Determine based on services provided, see below. Max Duration: Type the number of seconds in the Max Duration field that will designate the maximum allowable duration of a trunk-to-trunk call. If Max Duration is configured to 0 (zero), Sphericall places no limit on the call length. DEFAULT is 21600 (or 6 hours). The Max Duration edit box accepts integers between 0 (no limit) and 2147483647 seconds (24855 days). An example of a longer call that may be valid via specific trunks would be an ongoing call to a conference bridge. In Service: Default is ENABLED. Allow Emergency Calls from non-emergency group Stations: Default is ENABLED. The following information helps you to understand the provisioning of the Total Capacity of the SIP trunks as configured on this window:
Total Capacity total number of simultaneous calls the softtrunk can support
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the number of simultaneous inbound calls the administrator wants to accept. the number of simultaneous outbound calls the administrator wants to allow.
Guaranteed Outbound = Total Capacity - Inbound Calls Guaranteed Inbound = Total Capacity - Outbound calls Example Configuration: Total Capacity=10, Inbound=8, Outbound=7 would imply the following: Guarantee 3 inbound calls with max of 8 Guarantee 2 outbound calls with max of 7 5 slots can be either inbound or outbound
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Add any Outbound Caller IDs that are needed for your system and this softtrunk. Note: Only change these settings if you wish to block certain Caller IDs from being
sent out this trunk. This is an area that should not be changed unless there is an advanced reason to configure it.
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Some SIP trunk service providers require that authentication be turned on when communicating over a SIP trunk. This must be configured on the Authentication tab and the details will be provided by the service provider.
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NOTE: entries in these fields provide authentication. Contact your service provider for this information.
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By enabling this field, the far end SIP service provider authenticates calls. Account: Provided by the service provider. Think of this as a User Name field. Password: Provided by the service provider. Verify if adding or changing. Think of this as a Password field.
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Realm: Provided by service provider. Think of this as similar to a domain field. Type: Default type is MD5. Use only MD5 as the type in this field. Only MD5 is currently supported. Authorization Type: It is important to note that in SIP, registration is used for routing incoming SIP requests and has no role in authorizing outgoing requests. Authorization and authentication are handled in SIP on a request-by-request basis with a challenge/response mechanism. Together, the account, password and realm form the credentials needed to access a SIP trunk. Individually, these fields are arbitrary. When a SIP trunk challenges a request from the MGC, the trunk sends the realm and a random value called a nonce to the MGC. The MGC must then resubmit the request with an authorization header that contains a hash. The hash is computed using the account, password, realm and nonce. A correct hash proves to the SIP trunk that the MGC knows the account and password. If the administrator is dealing with a SIP service provider, these credentials are specified by the service provider. The Sphericall system needs to have the SIP service providers Account (user name), Password (Password) and Realm (domain) in order for the Sphere system to either challenge, respond or both prior to accepting a call into the system. Both (Challenge and Respond): This system will challenge and respond to challenges. This authorization credential can be used both to challenge incoming INVITEs and to respond to outgoing INVITE and REGISTER challenges. To Challenge: This system will Challenge all incoming calls and will authenticate each call. This Authorization credential is to be used only to challenge the incoming INVITEs (Sphericall Manager is receiving the INVITE). To Respond: This system will respond to challenges. This Authorization credential is to be used only to respond to an INVITE or a REGISTER challenge (Sphericall Manager has sent the INVITE/REGISTER and the destination is challenging). Unknown: This means no authentication is to be used in either direction (Incoming/Outgoing). Authentication might be set to UNKNOWN to disable the authentication temporarily (or another option is to uncheck the enable/disable box, which loses the authentication details).
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Select Add to add a DID Mapping to the DID Mappings area (optional). Select Add to Schedule the Default Routing (optional). Enter a DID in the Input DID and select Test to verify DID (optional). Click Apply. Select Outward Routing.
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Click Add Outside Service to add an outside service function to this softtrunk. Click Add to add Outside Dialing Rules if required.
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Dialing Rules are covered in more detail in Book 2: Install & Configure the Sphere System.
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Only add Emergency Groups or Settings if you are completing the Emergency Groups installation. Refer to Book 6: Emergency Service Installations.
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Select Settings.
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Setting can be set at the system level and at the trunk level. This setting redefines the SIP Timer B for Invite Transactions. Currently its being used only for INVITE client transactions where more than one destination IP address is available (DNS SRV records). 8000 milliseconds (default); Range: 5000 - 64000 This setting defines the percentage expiry after which the REGISTER requests to refresh the registration bindings should be sent. E.G. if the Registration refresh period is 1800 seconds and the threshold is set to 75%, then MGC will send a REGISTER request to refresh the binding every 1350 seconds. Value 50 (default) The following is a description of the method for putting a call on hold as defined in RFC 2543. If a party in a call wants to put the other party "on hold", i.e., request that it temporarily stops sending one or more media streams, a party re-invites the other by sending an INVITE request with a modified session description. The session description is the same as in the original invitation (or response), but the "c" destination addresses for the media streams to be put on hold are set to zero (0.0.0.0). This setting might need to be set to False for certain SIP trunk service providers.
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This scenario of a SIP tie line solution has the following characteristics: This is the essence of soft trunking. When you set up the system appropriately, and assign a softtrunk using only an IP connection, two different organizations may use SIP for control of calls to each other, as well as sending calls out to the PSTN. Companies may use SIP for their call control over their WAN calls, but may still use other media gateways to access the PSTN.
YOU MUST FIRST VERIFY OR ADD THIS SIP ENDPOINT AS A USER AGENT TO THE SYSTEM. Refer to the beginning of this chapter for this procedure for all SIP endpoints. Refer to Book 1: Planning & Preparing the Sphere System for information on ordering and planning SIP services. Once you have your IP/internet services delivered and the Customer Premise Equipment (CPE) is onsite (typically an ALGApplication Layer Gateway), connect the CPE to the local area network (straight-through RJ-45 cable from the CPE to the Ethernet network switch). During the planning process of your SIP tie line, you must secure the following information:
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Account Number Service Provider Domain name Outbound Proxy Registration Type Contact Domain Primary MGC (Sphericall Manager) Secondary MGC (Sphericall Manager) Password Realm
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In some cases, the service provider may only make changes or additions to your services or the service order through one individual.
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Review the User Agents listed in the window. If the SIP endpoint you are using is not listed in this window, you must add it. Click Add. Enter the User Agent name. Select Endpoint Type: REQUIRED. The Sphere system requires this information in order to know how to treat it or what features to apply per endpoint. Enter an Agent Description that is appropriate for the Name & Endpoint Type. Click Apply. Note: Those User Agents listed in the SIP dialog window that also have the Default
checked, are those User Agents created into the system by default. These will remain in the system. User Agents added by system administrators are not indicated with a check in the Default column.
From the Sphericall Administration application: Select the Trunks tab. Right-click and select Add. Choose Add Softtrunk. Fill in the fields as follows:
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NOTE: Information entered here will reflect the OTHER Sphere systems information.
Tie Lines Between Sphericall Systems Since the time to failover is determined by the registration lifetime of inbound registrations, we recommend that tie lines between Sphericall systems be configured as registration type None. However, it is still possible to use inbound/outbound registration by configuring one end of the tie line for Inbound registration and the other end for Outbound registration. In this case, both ends of the tie line must be configured with the same account and password on the Authorization tab of the Trunk properties dialog box. This is required since either side of the tie line may challenge the other, and we do not support "inbound" and "outbound" credentials. Type in the information for each field:
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Description: Any printable ASCII character except double quote or backslash. This field is a description field that specifies the display-name in the From and To headers of Outbound REGISTER requests. Often it will be the name that is assigned to this service Example: LincolnWashingtonTieLine (noting the two systems you are tying together). Account: Information regarding this tie line (i.e. could be the same info as entered in the Description field for this tie line). Example: LincolnWashingtonTieLine (noting the two systems you are tying together). The Account field specifies the userinfo portion of the SIP URI in the From and To headers of outbound REGISTER requests. The Account field may contain any printable character except the following: space, pound sign #, percent sign %, colon :, less than <, greater than >, at sign @, square brackets [], backslash \, caret ^, brace brackets {}, pipe | and grave accent `. Service Provider Domain: Information in this field will represent WHO you are sending your registration TO. For the tie line, you fill in the info of the OTHER Sphere system in this field. Example: Lincoln.sip.company.net The Service Provider Domain field (SPD) may be filled with an IP address, DNS host name or DNS domain name. The SPD specifies the hostname portion of the SIP URI in the From and To headers of outbound REGISTER requests. The SPD also specifies the destination for outbound SIP requests if the Outbound Proxy (OP) is blank. A DNS lookup is performed on the
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contents of the SPD if it does not contain an IP address. The type of DNS lookup performed is specified by the DnsNaptrServiceEnabled and DnsSrvServiceEnabled settings (see Note below).
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Outbound Proxy: Recommend using an IP Address of the other Sphere MGC on the other end of the tie line. Example: 66.237.127.53 The Outbound Proxy field (OP) used to be called the Host field. The OP may be filled with an IP address, DNS host name, DNS domain name or it may be left empty. If the OP is not empty, all outbound SIP requests are sent to the specified destination. A DNS lookup is performed on the contents of the OP if it does not contain an IP address. The type of DNS lookup performed is specified by the DnsNaptrServiceEnabled and DnsSrvServiceEnabled settings (see Note below). If the OP is empty, the destination for outbound SIP requests is determined by the Service Provider Domain field.
Summary The SPD and OP determine where the Sphericall system sends outbound SIP requests. The OP field would typically be used when the administrator wants to force all outbound SIP messages through a specific device, for example, a SIP Session Border Controller (SBC).
Note: The following two System Initialization Settings could affect settings for the
Port information: use the default. Registration type: Recommended for Tie Line installations: NONE
Inbound - the SIP trunk sends a register request to the Sphericall system. When the SIP trunk first initiates contact by sending a REGISTER request, the MGC automatically creates a hub, trunk and outside service. The Service Provider tab does not appear in the Hub device properties dialog box since the SIP trunk provides all the required information in the REGISTER request. Outbound - the Sphericall system sends REGISTER request to the SIP trunk; all fields on the Service Provider tab (trunk device properties) are used. None - issues no registration; neither the SIP trunk nor the Sphericall system send a REGISTER request; all fields on the Service Provider tab except the Contact Domain are used (trunk device properties). Primary MGC:
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Select the Media Server (MGC) name from the drop down box for the Primary. Example: Lincoln1 would be selected
Secondary MGC:
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Leave the drop down box set to NONE for the Secondary.
The Primary and Secondary MGC fields specify which MGCs are primary and secondary for the SIP trunk. The Secondary MGC assumes control of the trunk should the Primary fail. For tie line installations, with registration type None, and no information in the Contact Domain, set the Primary to reflect the Media Server (MGC) name. The following information about Primary and Secondary is accurate, but used only if the Contact Domain information field is used, which is rarely used for tie line implementations, but may occasionally be used if the tie is installed with an Outbound REGISTER. If the CD specifies an IP address or DNS host name, the Secondary MGC field should be filled with "None." In this case, specifying a Secondary MGC has limited value since the external system will not be able to locate the Secondary MGC to send SIP requests to. In practice, calls inbound to the Sphericall system will fail. Calls outbound from the Sphericall system may fail. If the CD specifies a DNS domain name, the DNS SRV records must match the contents of the Primary and Secondary MGC fields. Specifically, the "first" SRV record must point to the Primary MGC and the "second" SRV record must point to the Secondary MGC. The order in which the SRV records are returned are determined by the priority and weight values assigned to the SRV records in DNS. The external system must attempt to contact the MGC pointed to by the SRV record with the lowest priority first. If multiple SRV records have the same priority, the external system must attempt to contact the MGC pointed to by the SRV record with the highest weight first. Since Microsoft's DNS server returns different orderings for SRV records with the same priority from query to query, we recommend setting the (priority, weight) of the SRV record for the Primary and Secondary MGC to (0, 0) and (1000, 0) respectively. If the CD specifies a DNS domain name, that name must be resolvable from the external system. For example, we cannot use internaldomain.spherecom.com as the CD for Cbeyond sinceinternaldomain.spherecom.com cannot be resolved anywhere other than the internal Sphere network.
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Contact Domain: For Tie Line installations, when using NONE for the registration type, Contact Domain information will not be filled in. How do the other side contact you? (IP address or FQDN): The Contact Domain field (CD) specifies the hostname portion of the SIP URI in the Contact header of outbound REGISTER requests. In other words, the CD determines how the external system locates an MGC to send SIP requests to. The CD may be filled with an IP address, DNS host name or DNS domain name. If filled with an IP address or host name, the external system will be bound to a single MGC with no redundancy. The CD should only be filled with a DNS domain name if the external system supports the DNS NAPTR and SRV lookup procedures specified by RFC 3263. Notice that the hostname portion of the SIP URI in the Contact header of all outbound non REGISTER requests is filled with the IP address of the MGC sending the request. This binds the SIP dialog to the MGC. This is necessary because we do not share call state information between MGCs. That is, it is not possible for a second MGC to assume control of a call in the middle of the call in the case of an MGC failure.
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Typically UDP. This determines the preferred network transport protocol for SIP signaling (TCP, TLS, UDP (default)). Note: For SIP trunks that use registration types Outbound or None, this setting specifies what transport is used to send SIP requests such as REGISTER and INVITE to the SIP Trunk.
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Sphere has included for selection most of the common types of User Agents. If the user agent your system is using is not offered here, you must create it in the System General Properties (SIP) window, then apply it to the User Agent associated with the trunk.
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Click OK.
The creation of the softtrunk is complete. SIP Trunk MAC Address When a trunk checks into a Sphericall system for the first time, the MGC automatically creates the necessary database records for it. As part of the record creation process, the MGC assigns a unique name to the trunk. For Sphericall MGs, the MGC uses the MG's MAC address as the unique name. For SIP trunks, the MAC address assigned to the trunk depends upon the registration type. For outbound registrations and registrations of type none, the MGC uses "Account@Service Provider Domain." For inbound registrations, the MGC uses the SIP URI of the REGISTER request To header. It is important to note that if the URI changes, the MGC will create new records for the trunk the first time it sends a REGISTER with the new URI.
Figure 17.14 Softtrunk hub properties
Once creation of the SIP softtrunk tie line is complete, the following information will be available on the Hub Properties:
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Name: Name of device (can be renamed for user-friendliness). MAC Address: Assigned to the device by the system. Stations/Trunks: Information on number of stations or trunks. LAN: Hub belongs to LAN displayed in this field.
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Localization Settings: Country/local language settings based on Localization Settings profile. Description: Information in this field can be edited. Hub Number: Automatically numbered by the system (recommended that this number remains unchanged). Last Check In Time: Information is updated upon each check-in. Firmware Version: None. Device Type: Default device type for Softtrunk is Unknown SIP Agent.
Name: Type name of softtrunk Hardware ID: Usually assigned dynamically by the system. Telephony Area: Assign or change Telephony area here. Zone: Assign or change Zone here. Total Capacity: Configure based on services provided by SIP provider. Inbound Capacity: Determine based on services provided. Outbound Capacity: Determine based on services provided.
Note: Administrators can shape the inbound and outbound activity on this softtrunk.
Max Duration: Type the number of seconds in the Max Duration field that will designate the maximum allowable duration of a trunk-to-trunk call. Default = 0 = No Limit. In Service: Default is ENABLED.
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Add any Outbound Caller IDs that are needed for your system and this softtrunk. Click Apply. Select the Authorization tab.
Entries here must match the other end of the tie line.
By enabling this field, the far end SIP server authenticates calls. Tie lines currently have two restrictions (please see restrictions below). Account: Provided by the other Sphericall Manager. Think of this as a User Name field. Password: Provided by the other Sphericall Manager. Verify if adding or changing. Think of this as a Password field. Realm: Provided by the other Sphericall Manager. Think of this as similar to a domain field. Type: Default type is MD5. Use only MD5 as the type in this field. Authorization Type: It is important to note that in SIP, registration is used for routing incoming SIP requests and has no role in authorizing outgoing requests. Authorization and authentication are handled in SIP on a request-by-request basis with a challenge/response mechanism. Together, the account, password and realm form the creditials needed to access a SIP trunk. Indiviually, these fields are arbitrary
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When a SIP trunk challenges a request from the MGC, the trunk sends the realm and a random value called a nonce to the MGC. The MGC must then resubmit the request with an authorization header that contains a hash. The hash is computed using the account, password, realm and nonce. A correct hash proves to the SIP trunk that the MGC knows the account and password. If the administrator is dealing with a SIP service provider, these credentials are specified by the service provider. Restrictions: If the administrator is setting up a SIP tie line between two Sphericall systems, there are two restrictions: The credentials must be identical on both ends since we only support one set of credentials per trunk. That is, we cannot challenge with one set of credentials and respond with a second set. Currently, for tie lines, the realm must be set to Sphericall.
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Select Add to add a DID Mapping to the DID Mappings area (optional). Select Add to Schedule the Default Routing (optional). Enter a DID in the Input DID and select Test to verify DID (optional). Click Apply. Select Outward Routing.
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Click Add Extension or Tie Line to add this function to this softtrunk. Click Add to add Outside Dialing Rules if required. Click Apply. Click OK.
Only add Emergency Groups or Settings if you are completing the Emergency Groups installation. Refer to Book 6: Emergency Service Installations.
Select the Trunks tab from the main window (this tab should be BLUE to indicate a new device). Double-click to select the new softtrunk device. Check the entries for the device properties. Note: There will be no MGCs tab when using softtrunk tie lines.
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Verify the Service Provider information (optional). Double-click to open the softtrunk trunk properties. Enter general information as needed on the General tab. Select the Authorization tab. Enter into these fields the SAME information that you would find in these fields at the other end of the tie line. Complete the fields for this authorization as follows:
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NOTE: Authorization is optional for tie line implementations, based on how you wish to communicate with the other Sphere system that you are tie-ing to; entries in these fields must match entries in the Authorization tab at the other side of the tie line.
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By enabling this field, the far end SIP server authenticates calls. Account: Provided by the other Sphericall Manager. Think of this as a User Name field. Password: Provided by the other Sphericall Manager. Verify if adding or changing. Think of this as a Password field. Realm: Provided by the other Sphericall Manager. Think of this as similar to a domain field. Type: Default type is MD5. Use only MD5 as the type in this field. Authorization Type: It is important to note that in SIP, registration is used for routing incoming SIP requests and has no role in authorizing outgoing requests. Authorization and authentication are handled in SIP on a request-by-request basis with a challenge/response mechanism. This system needs to have the SIP servers Account (user name), Password (Password) and Realm (domain) in order for the Sphere system to either challenge, respond or both prior to accepting a call into the system. Both (Challenge and Respond): This system will challenge and respond to incoming calls. This authorization credential can be used both to challenge incoming INVITEs and to respond to outgoing INVITE and REGISTER challenges To Challenge: This system is set to Challenge all incoming calls and will authenticate each call. This Authorization credential is to be used only to challenge the incoming INVITEs (MGC is receiving the INVITE). To Respond: This system is set to respond to authentication of the call as it comes in. This Authorization credential is to be used only to respond to an INVITE or a REGISTER challenge (MGC has sent the INVITE/REGISTER and the destination is challenging). Unknown: This means no authentication is to be used in either direction (Incoming/Outgoing). Authentication might be set to UNKNOWN to disable the authentication temporarily (or another option is to uncheck the enable/disable box, which loses the authentication details).
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Click Apply. Complete the configuration of Inbound, Outbound and Default Routing as desired.
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This example scenario of a SIP tie line to call center solution has the following characteristics: Sphere system administrators are able to connect to a call center for call queuing, processing, etc. Calls are directed from the Sphere system to the call center server/manager via DID, Auto Attendant routing, default route, etc. This tie-line connection method can be used for a voice mail setup, fax server, call center
YOU MUST FIRST VERIFY OR ADD THIS SIP ENDPOINT AS A USER AGENT TO THE SYSTEM. Refer to the beginning of this chapter for this procedure for all SIP endpoints. Refer to Book 1: Planning & Preparing the Sphere System for information on planning SIP services. Once you have your SIP services delivered and the Customer Premise Equipment (CPE) is onsite (typically an ALGApplication Layer Gateway), connect the CPE to the network (straight-through RJ-45 cable from the CPE to the network switch). During the planning process of your SIP service, you must secure the following information:
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Account Number Service Provider Domain name Outbound Proxy Registration Type Contact Domain Primary MGC (Sphericall Manager) Secondary MGC (Sphericall Manager) Password Realm
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To configure SIP
From the Sphericall Administration application:
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Select the Trunks tab. Right-click and select Add. Choose Add Softtrunk. Fill in the fields as follows:
Description: Any printable ASCII character except double quote or backslash. This field is a description field that specifies the display-name in the From and To headers of Outbound REGISTER requests. Often it will be the phone number that is assigned to this service. Example: 8477932222 Account: Provided to you by call center service provider (i.e. could also be phone number that is assigned to this service). Example: 8477932222 The Account field specifies the userinfo portion of the SIP URI in the From and To headers of outbound REGISTER requests. The Account field may contain any printable character except
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the following: space, pound sign #, percent sign %, colon :, less than <, greater than >, at sign @, square brackets [], backslash \, caret ^, brace brackets {}, pipe | and grave accent `.
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Service Provider Domain: Provided by the call center service provider Example: 777445.sip.company.net. The Service Provider Domain field (SPD) may be filled with an IP address, DNS host name or DNS domain name. The SPD specifies the hostname portion of the SIP URI in the From and To headers of outbound REGISTER requests. The SPD also specifies the destination for outbound SIP requests if the Outbound Proxy (OP) is blank. A DNS lookup is performed on the contents of the SPD if it does not contain an IP address. The type of DNS lookup performed is specified by the DnsNaptrServiceEnabled and DnsSrvServiceEnabled settings (see Note below). Outbound Proxy: Provided by the call center service provider Example: siptrunkinfo.sip.callcenterserver.net The Outbound Proxy field (OP) used to be called the Host field. The OP may be filled with an IP address, DNS host name, DNS domain name or it may be left empty. If the OP is not empty, all outbound SIP requests are sent to the specified destination. A DNS lookup is performed on the contents of the OP if it does not contain an IP address. The type of DNS lookup performed is specified by the DnsNaptrServiceEnabled and DnsSrvServiceEnabled settings (see Note below). If the OP is empty, the destination for outbound SIP requests is determined by the Service Provider Domain field.
Summary The SPD and OP determine where the Sphericall system sends outbound SIP requests. The OP field would typically be used when the administrator wants to force all outbound SIP messages through a specific device, for example, a SIP Session Border Controller (SBC).
Note: The following two System Initialization Settings could affect settings for the
Port information: use the default. Registration Type: Recommended for softtrunk to call center: OUTBOUND
Inbound - the SIP trunk sends a register request to the Sphericall system. When the SIP trunk first initiates contact by sending a REGISTER request, the MGC automatically creates a hub, trunk and outside service. The Service Provider tab does not appear in the Hub device properties dialog box since the SIP trunk provides all the required information in the REGISTER request.
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Outbound - the Sphericall system sends register request to the SIP trunk; all fields on the Service Provider tab (trunk device properties) are used. None - issues no registration; neither the SIP trunk nor the Sphericall system send a register request; all fields on the Service Provider tab except the Contact Domain are used (trunk device properties).
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Primary MGC:
Secondary MGC: The Primary and Secondary MGC fields specify which MGCs are primary and secondary for the SIP trunk. The Secondary MGC assumes control of the trunk should the Primary fail. If the CD specifies an IP address or DNS host name, the Secondary MGC field should be filled with "None." In this case, specifying a Secondary MGC has limited value since the external system will not be able to locate the Secondary MGC to send SIP requests to. In practice, calls inbound to the Sphericall system will fail. Calls outbound from the Sphericall system may fail. If the CD specifies a DNS domain name, the DNS SRV records must match the contents of the Primary and Secondary MGC fields. Specifically, the "first" SRV record must point to the Primary MGC and the "second" SRV record must point to the Secondary MGC. The order in which the SRV records are returned are determined by the priority and weight values assigned to the SRV records in DNS. The external system must attempt to contact the MGC pointed to by the SRV record with the lowest priority first. If multiple SRV records have the same priority, the external system must attempt to contact the MGC pointed to by the SRV record with the highest weight first. Since Microsoft's DNS server returns different orderings for SRV records with the same priority from query to query, we recommend setting the (priority, weight) of the SRV record for the Primary and Secondary MGC to (0, 0) and (1000, 0) respectively. If the CD specifies a DNS domain name, that name must be resolvable from the external system. For example, we cannot use internaldomain.spherecom.com as the CD for Cbeyond sinceinternaldomain.spherecom.com cannot be resolved anywhere other than the internal Sphere network.
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Contact Domain: You provide this information Example: 11.2.27.122 The Contact Domain field (CD) specifies the hostname portion of the SIP URI in the Contact header of outbound REGISTER requests. In other words, the CD determines how the external system locates an MGC to send SIP requests to. The CD may be filled with an IP address, DNS host name or DNS domain name. If filled with an IP address or host name, the external system will be bound to a single MGC with no redundancy. The CD should only be filled with a DNS domain name if the external system supports the DNS NAPTR and SRV lookup procedures specified by RFC 3263. Notice that the hostname portion of the SIP URI in the Contact header of all outbound non REGISTER requests is filled with the IP address of the MGC sending the request. This binds the SIP dialog to the MGC. This is necessary because we do not share call state information between MGCs. That is, it is not possible for a second MGC to assume control of a call in the middle of the call in the case of an MGC failure. Preferred Transport: Select the appropriate Transport means.
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Typically UDP. This determines the preferred network transport protocol for SIP signaling (TCP, TLS, UDP (default)). Note: For SIP trunks that use registration types
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Outbound or None, this setting specifies what transport is used to send SIP requests such as REGISTER and INVITE to the SIP Trunk.
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Sphere has included for selection most of the common types of User Agents. If the Call Center User Agent your system is using is not offered here, you must create it in the System General Properties (SIP) window, then apply it to the User Agent associated with the trunk.
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Click OK.
The creation of the softtrunk is complete. SIP Trunk MAC Address When a trunk checks into a Sphericall system for the first time, the MGC automatically creates the necessary database records for it. As part of the record creation process, the MGC assigns a unique name to the trunk. For Sphericall MGs, the MGC uses the MG's MAC address as the unique name. For SIP trunks, the MAC address assigned to the trunk depends upon the registration type. For outbound registrations and registrations of type none, the MGC uses "Account@Service Provider Domain." For inbound registrations, the MGC uses the SIP URI of the REGISTER request To header. It is important to note that if the URI changes, the MGC will create new records for the trunk the first time it sends a REGISTER with the new URI.
Figure 17.21 Softtrunk hub properties
Once creation of the SIP softtrunk tie line is complete, the following information will be available on the Hub Properties:
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Name: Name of device (can be renamed for user-friendliness). MAC Address: Assigned to the device by the system. Stations/Trunks: Information on number of stations or trunks. LAN: Hub belongs to LAN displayed in this field. Localization Settings: Country/local language setting based on the Localization Setting profile. Description: Information in this field can be edited.
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Hub Number: Automatically numbered by the system (recommended that this number remains unchanged). Last Check In Time: Information is updated upon each check-in. Firmware Version: None. Device Type: Default device type for Softtrunk is Unknown SIP Agent.
Name: Type name of softtrunk Hardware ID: Usually assigned dynamically by the system. Telephony Area: Assign or change Telephony area here. Zone: Assign or change Zone here. Total Capacity: Configure based on information from the call center application. Inbound Capacity: Configure based on information from the call center application. Outbound Capacity: DConfigure based on information from the call center application.
Note: Administrators can shape the inbound and outbound activity on this softtrunk.
Max Duration: Type the number of seconds in the Max Duration field that will designate the maximum allowable duration of a trunk-to-trunk call. In Service: Default is ENABLED. Emergency Calls Enabled: Default is ENABLED.
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The following information helps you to understand the provisioning of the Total Capacity of the SIP trunks as configured on this window:
Total Capacity Inbound Capacity Outbound Capacity total number of simultaneous calls the softtrunk can support the number of simultaneous inbound calls the administrator wants to accept. the number of simultaneous outbound calls the administrator wants to allow.
Guaranteed Outbound = Total Capacity - Inbound Calls Guaranteed Inbound = Total Capacity - Outbound calls Example Configuration: Total Capacity=10, Inbound=8, Outbound=7 would imply the following: Guarantee 3 inbound calls with max of 8
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Add any Outbound Caller IDs that are needed for your system and this softtrunk. Click Apply. Select the Authorization tab.
NOTE: This field is optional for tie line implementations, based on how you wish to communicate with the call center system that you are tie-ing to: entries in these fields indicate who THIS trunk is on this side of the tie line.
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By enabling this field, the far end SIP call center authenticates calls. Account: Provided by the call center system. Think of this as a User Name field.
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Password: Provided by the call center system. Verify if adding or changing. Think of this as a Password field. Realm: Provided by call center system. Think of this as similar to a domain field. Type: Default type is MD5. Use only MD5 as the type in this field. Authorization Type: It is important to note that in SIP, registration is used for routing incoming SIP requests and has no role in authorizing outgoing requests. Authorization and authentication are handled in SIP on a request-by-request basis with a challenge/response mechanism. This system needs to have the SIP call centers Account (user name), Password (Password) and Realm (domain) in order for the Sphere system to either challenge, respond or both prior to accepting a call into the system. Both (Challenge and Respond): This system will challenge and respond to incoming calls. This authorization credential can be used both to challenge incoming INVITEs and to respond to outgoing INVITE and REGISTER challenges To Challenge: This system is set to Challenge all incoming calls and will authenticate each call. This Authorization credential is to be used only to challenge the incoming INVITEs (MGC is receiving the INVITE). To Respond: This system is set to respond to authentication of the call as it comes in. This Authorization credential is to be used only to respond to an INVITE or a REGISTER challenge (MGC has sent the INVITE/REGISTER and the destination is challenging). Unknown: This means no authentication is to be used in either direction (Incoming/Outgoing). Authentication might be set to UNKNOWN to disable the authentication temporarily (or another option is to uncheck the enable/disable box, which loses the authentication details).
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Select Add to add a DID Mapping to the DID Mappings area (optional). Select Add to Schedule the Default Routing (optional). Enter a DID in the Input DID and select Test to verify DID (optional). Click Apply. Select Outward Routing.
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Click Add to add Outside Dialing Rules if required. Click Apply. Click OK.
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Session Initiation Protocol, SIP, is a protocol for transporting call setup, routing, authentication and other feature messages to endpoints within the IP domain. Within the Sphere system, SIP is used to allow external systems to participate in calls with the Sphericall Manager. The Manager targets the use of SIP for integration with some specific third-party products for integration with the following: two-way calls (Sphericall Manager and external system), calls between two systems placed on hold, transfer of calls between the two systems, passing of DTMF digits into the thirdparty system during a call, and notification of message waiting. The Terminal location logic to accommodate gateways that register multiple trunks from the same IP address. Understanding this logic will help enormously when troubleshooting SIP connection issues. The Sphericall Manager (MGC) uses the following logic to locate a SIP terminal in 6.0. and later: From Header Userinfo is compared to the Account field of the Service Provider information. If no match is found, the Sphericall Manager moves to step 2. Note: this is the most common way stations are identified, but does not help for trunks since the FROM field contains the caller ID of the incoming call. To Header Userinfo is compared to the DID maps configured for SIP trunks. If two trunks have overlapping DID maps, the Sphericall Manager moves to step 4. If no match is found, the Sphericall Manager moves to step 3. Contact Header hostname:port is compared to the Outbound Proxy if configured, otherwise the Service Provider Domain of the Service provider information. If the hostname:port is an IP address it is compared exactly to what is configured. If the hostname:port is not an IP address, a partial compare is performed against the Service Provider information. For example, the hostname "horatio.rndlab.spherecom.com" would match the Service Provider information "rndlab.spherecom.com". Note: in both cases the port must match. If more than one User Agent matches this criterion, or no match is found, the Sphericall Manager moves to step 4. Authorization Header If the request contains an Authorization header, the credentials included in the Authorization Header are compared against the credentials configured in the Sphericall Admin Authorization window. If no match is found, the Sphericall Manager moves to step 5.
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The Sphericall Manager (MGC) challenges the sender to obtain credentials via the Authorization Header.
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OVERVIEW
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The Quintum Tenor AF series analog VoIP gateways are ideal for small Sphericall systems or small office locations that are part of a larger distributed Sphericall system. The Quintum Tenor AF series analog VoIP gateways will offer enhanced capabilities for Sphericall users in two key areas. First they offer a flexible and cost effective small office and remote office options for both analog trunk and station connections. The Quintum gateways offer a range of options for trunk, station and combined gateway solutions, in a small, cost effective foot print, that is certified to operate with Sphericall. Second, they also offer additional small / remote site survivability options for Sphericall users. By leveraging SIP as an open standard, and the Quintum gateways built in proxy and stand-by call agent, Sphericall users can leverage the combined Sphere and Quintum solution to further increase the high availability capabilities of a Sphericall system. The powerful combination of the Sphericall IP PBX and Quintum Tenor gateways provides a complete Unified Communications solution that delivers communications services across your entire enterprise, even at the smallest of office locations. When using the Quintum gateway as part of a survivability solution, the Quintum gateway is configured as an Outbound SIP Proxy on behalf of the SIP telephones at a remote office site. As an Outbound SIP Proxy, the Quintum gateway remains in the signaling path between the SIP telephones and a Sphericall Manager that is located across a customer WAN. The Quintum gateway maintains a registration with the Sphericall Manager. If the connection to the Sphericall Manager is interrupted (for instance, a WAN failure), the Quintum gateway will assume responsibility for handling call processing requests for the SIP telephones that are configured to use it as an Outbound SIP Proxy. This includes routing calls between any device (SIP Phone, analog phone, analog trunk) that is connected to the Quintum gateway. Following are some examples: Example 1: A local SIP Phone places a local PSTN call during a WAN failure A SIP phone users dials the outside service digit 8 and the 11 digit PSTN number 1-555-555-5555.
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Overview
The Quintum Outbound SIP Proxy attempts to forward the signaling messages to the Sphericall Manager. Upon recognizing that the Sphericall Manager is not available, the Quintum gateway will look at the SIP messages and match the dialed number to its internal dial plan rules which instructs the Quintum gateway to route calls with an outside service prefix to the local PSTN trunks after removing the prefix from the digit string. Example 2: Inbound call routing from the PSTN during a WAN Failure For inbound calls over analog trunks, most often these calls are routed by default to a Sphericall Manager auto-attendant. In this environment, the Sphericall Manager is reachable at another location across a WAN. If the WAN fails, and the Sphericall Manager is not reachable, then the Quintum Tenor can route inbound PSTN calls to one of the FXS ports or a SIP phone. The following Quintum Tenor gateway models are certified for use with Sphericall. Note that each Quintum Tenor Survivable gateway listed below can support up to 50 SIP telephones as an Outbound SIP Proxy.
Table 18.1 Quintum Gateway models
Tenor AF Series Gateways
AFT200: 2 FXO ports AFT400:4 FXO ports AFT800: 8 FXO ports AFG200: 2 FXO ports AFG400:4 FXO ports AFG800: 8 FXO ports AFE400:4 FXS + 2 FXO ports AFE600: 6 FXS +2 FXO ports AFM200: 2 FXS +2 FXO ports AFM400: 4 FXS +4 FXO ports Power Supply Power Cable
AF Classic
501-1199-00 501-1202-00 501-1195-00 501-1198-00 501-1201-00 501-1194-00 501-1203-00 501-1205-00 501-1200-00 501-1204-00 Universal North America
AF Survivable
501-1199-SG 501-1202-SG 501-1195-SG 501-1198-SG 501-1201-SG 501-1194-SG 501-1203-SG 501-1205-SG 501-1200-SG 501-1204-SG 520-1033-00 501-1190-00
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Operation
OPERATION
Figure 18.1 Quintum Solutions
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Quintum gateways provide options for supporting analog stations and trunks on the Sphere system. Once configured, the Sphericall Manager (MGC) has full control of calls placed and received on the phones and trunks connected to the Quintum gateway. If using a survivable Quintum gateway, the Quintum gateway knows about the IP and analog phone activities. If at some time the phone sends a signal intended for the Sphericall Manager, and the WAN connection is not available, the Quintum gateway will retry the Sphericall Manager at least 3 times (this is configurable). If there is no reply within the specified time period, the gateway will attempt to complete the call itself. If the call was placed to another extension at that same remote site, it will transfer the call locally. If the call was placed to the PSTN, the gateway will dial the PSTN and pass the call through to the PSTN locally. The Quintum gateway cannot, however, pass the call to another networked location, since the WAN connection is not available. Example: If you dial 8-1-847-793-9600 from the Quintum side, the Quintum will offer the call to the Sphericall Manager first. If the Sphericall Manager is down, it consults its internal routing tables. In the hopoff directory you'd have an entry that says anything that starts with 81, strip off the 8 and send it to an FXO port. Example: Local PSTN numbers can be routed directly in to the Quintum FXO/FXS unit and configured to first go to the Sphericall Manager auto attendant. If there is a WAN failure, the call would go to one FXS phone on the Quintum side.
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Planning
PLANNING
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Verify Sphere System Requirements for Quintum gateways version for interoperability with Sphericall. Specific versions of Quintum are only compatible with specific Sphericall versions. The Sphere system should be installed, configured and tested as fully functional. Refer to the Quintum VoIP Gateway User Manual for installation planning, setup, package contents, safety, and conditions of use. Quintum units can be used on local sites as diagramed above.
PREPARING
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Set up Quintum unit as directed by package instructions. Appropriate power supply and power cord for each unit and country should be verified prior to installation. Prepare network according to Sphere System Requirements. Ensure that you have the latest Quintum software loaded using the manufacturers instructions.
www.quintum.com/support
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If you are using a survivable gateway, you must update the applicable license file.
The license file will only work on the Quintum device with the serial number that matches the one provided when the license file was generated. The license file is not interchangeable among devices. Consult Quintum at: www.quintum.com/support.
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Know the required login for working with this gateway: Default Username: admin Default Password: admin
Connect appropriate LAN cable to the LAN port. Connect RJ-45 straight-through pinout cable for PSTN connection to the appropriate port. Connect RJ-11 cable(s) for telephone connection to the appropriate port(s)analog only. Connect SIP phones to appropriate network port(s). Connect the standard RS-232 serial cable to the Console Port. Connect AC Power Adapter jack to the power supply and to the power source. Be sure to have the Product Guide for the appropriate Tenor products in CD form.
There is a version of the Configuration Manager on the CD that is shipped with each product. Administrators should be sure to match the version of the Configuration Manager to the unit they are configuring as well as to the firmware version of the software on the gateway unit.
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Know the dial plan for all the extensions on the remote office node prior to installation of the Quintum gateway unit.
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Installing
Connect one end of the RJ-11 into the phone. Connect the other end of the RJ-11 into the FXS port on the Tenor device.
Connect one end of the SIP phone cable into the network/switch. Connect the Tenor device to the network.
INSTALLING
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There is more than one way to integrate the Quintum VoIP Gateway to the Sphericall system as a remote office communications device depending on the model and the network.
Connect the Quintum gateway to the network. It should obtain a DHCP IP address. Please see the Quintum manual for assigning a static IP address. Load the Quintum Configuration Management software to your Sphere Manager. Once completed, run the program. The Quintum Configuration Management software has the capability to discover (within the same subnet) all Quintum devices. Once the device is discovered, you can start the configuration by connecting to the device.
IP Address Configuration
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Installing
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11-digit Dialing:
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Dial Plan Country: None Progress Tone Country: USA, Canada, Mexico (should match your country) Country Code: 1 Area/City Code: Enter area code Long Distance Prefix: 1 Int'l Prefix: 0 Confirm/OK
Disconnect Generation -> Battery Removal Caller ID Generation -> FSK Phone Number/Extension -> Assign an extension to each channel. This is the same extension that is assigned in the Sphericall Administrator application (know your dial plan and extension assignment prior to configuring the Quintum gateway). Add
a. Number Pattern: Extension of phone assigned in Sphericall Admin b. Channel: Corresponding connection
Confirm/OK
Disconnect Detection -> Battery Removal Caller ID Detection -> FSK or DTMF Tone Based Answer Detection -> Disabled
Listening Port -> 5060 Local Gateway -> Yes Default Route Port -> 5061 Confirm/OK
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Installing
Channel Configuration
Disable Trunk/PSTN-side channels -> No
Configuration Summary
This dialog summarizes the configuration before writing it to the Quintum gateway. Once the Configuration Wizard is finished, further steps are required to configure the Quintum gateway for use with Sphericall. The CM has two modes of operation, basic and advanced.
VC-1: G.711 Mu-law Codec Payload Size: 20 ms VC-2: G.729AB Codec Payload Size: 20 ms Confirm/OK
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Installing
Sphericall classifies SIP User Agents as stations or trunks based upon the UserAgent header in the REGISTER request. Quintum allows the value of the UserAgent header to be specified. The required values for Sphericall are:
Quintum-FXS for FXS ports Quintum-FXO for FXO ports If running with Sphericall v5.2 (an early beta), these strings must also be entered into the SIP tab of the Sphericall Administrator System Properties dialog. Sphericall v6.0 will ship with these strings in the database.
Under VOIP Configuration, SIP Signaling groups, SIP signaling Group 1 Go to the Advanced Tab and change the User Agent Header to: Quintum-FXO/1.0.0.
This will allow a match of the name to the one in the Sphere system, and will allow the unit to check into Sphericall.
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Confirm/OK. A second SIP signaling group must be created to assign a different User Agent header to the FXS ports.
The FXS ports must be moved from the existing SIP signaling group to the new SIP signaling group.
To create a second SIP signaling group and move the FXS ports:
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Advanced Configuration. VoIP Configuration. SIP Signaling Groups. In the SIP Signaling Group-1 dialog, on the User Agent tab, delete the entries for the FXS ports. Right click on SIP Signaling Groups and select "New". Enter 2 as the group index. Select SIP Signaling Group-2 and enter a value for the Primary SIP Server. This will normally be the same as you configured for SIP Signaling Group-1. On the Advanced tab, change the User Agent Header to Quintum-FXS. On the User Agent tab, create entries for each FXS port, filling in the Primary User and Contacts[1] fields with the extension assigned to each FXS port in the Sphericall Administrator application.
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Installing
EXAMPLE 2 7 digit number Number Pattern: 7 Replacement: 7 Description: Local Type: Public
EXAMPLE 3 10 digit number [(xxx) 555-1212] Number Pattern: 3 Replacement: 3 Description: Local Type: Public
For 7- and 10-digit dialing, you must create an entry for each digit, 0-9.
Under Circuit Configuration/Signaling Configuration CAS Signaling Groups CAS Signaling Group-line on the General tab Set Signaling Type to Loop Start, Forward Disconnect. Set Forward Disconnect Delay to 200 milliseconds on the Signaling tab. Confirm/OK
Advanced Configuration.
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VOIP Configuration. IP Routing Group. IP Routing Group Default. Under the General Tab, uncheck the Silence Suppression box.
To enable Caller ID name display for calls inbound from the PSTN:
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Advanced Configuration. VOIP Configuration. IP Routing Group. IP Routing Group Default. Under the ANI/Fax Tab, go to the Relay Calling name field and select Relay CNAM in INVITE.
Advanced Configuration VOIP Configuration IP Routing Group IP Routing Group Default Under the Advanced tab, adjust the Rx Gain to increase the volume of outside parties as heard by inside parties Under the Advanced tab, adjust the Tx Gain to increase the volume of inside parties as heard by outside parties Note: Increasing the gains too much may lead to other undesirable effects such as
echo.
Advanced Configuration. Circuit Configuration. CAS Signaling Groups. CAS Signaling Group phone. Check Detect Flash Hook Signal. Set Maximum Flash-Hook Duration to 1100. Set Minimum Flash-Hook Duration to 350. Set Caller ID Generation to FSK.
To set the Quintum to pass Caller ID name to the Quintum FXS ports
Caller ID name is not set by default on the Quintum TENOR. Complete the following to change this default to pass Caller ID name.
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Advanced configuration. Circuit Configuration. CAS Signaling Groups. CAS Signaling Groups - Phone Signaling Tab. "Relay Calling Name" should be set to "Relay CNAM"
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Installing
To adjust the system for gateways that do not support T.38 Fax Relay
T.38 is enabled by default on Quintum gateways, however they do not fall back to G.711 if the other end of the call does not support T.38. If gateways that do not support T.38 are present in the system, the Fax Relay setting under Advanced Configuration VoIP Configuration IP Routing Groups IP Routing Group-default must be changed on the Fax/QOS tab from T.38 w/o fallback to T.38 w/G.711 X fallback, where X is Mu-law or A-law.
1 2 3 4 5 6
Advanced Configuration. VoIP Configuration. IP Routing Groups. IP Routing Group-default. Fax/QOS tab. Change T.38 w/o fallback to T.38 w/G.711 X fallback
Open Sphericall System Properties from the General Tab. Select the Media Streams tab. At the bottom of the dialog ensure the following setting: DTMF digit payload type RFC 2833: 101.
Trunk tab. Expand and find the trunk under the tree. Open the trunk properties. Verify that the trunk capacity, inbound and outbound is set to the number of trunks. Authorization tab: check Use Authorization. Enter Primary User: YY & Primary Password: ZZ Realm: Sphericall. Type: MD5 Authorization Type: To Respond/To Challenge
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Copy the firmware from the Sphericall CD and unzip the file (i.e. using a program such as WinZip). Save the files to a new, empty directory on your hard drive. From your PC, select Start> All Programs> Accessories> Command Prompt. The Command Prompt window is displayed. Use the CD command (cd\) to change to the directory containing the unzipped firmware files on your local hard drive. Type ftp followed by the IP address of the unit. Press Enter. Login with the username and password. Default for both is admin.
Type bin <Enter> Type hash <Enter> Type prompt <Enter> Type mput *.* <Enter>
6 7
Reset the gateway. To confirm the upgrade, initiate a telnet session with the unit and use the command show v.
...........................................................
Once the Quintum gateway is configured, it can be checked-in to the Sphericall Manager. Trunk and/or Station information should also be configured at the Sphericall
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Manager to set user address profiles, phone class of service profiles, trunk routing, etc. Performing the above documented configuration settings will assist you with general settings, please refer to Sphere documentation, specifically Book 2: Install & Configure for adding user and extension properties to the stations on the Quintum node.
...........................................................
Quintum has three preconfigured # codes to enable the features of hold and transfer calls. The default codes are:
Action
Hold a call Unhold a call
Feature Codes
#46 #46
Blind Transfer
#90 + extension
Attended Transfer
Flash Hook + Extension Once connected: Hangup to complete the transfer OR Announce the call and hangup OR #48 to cancel the transfer
feature management, any analog phones connected through the Quintum gateway are required to use the above feature # codes for features.
To configure codes
Open the advanced mode of the Configuration Manager under:
1 2
Circuit Configuration/Line Routing Configuration/Line Circuit Routing Group-phone, Call Services tab. Check the following:
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...........................................................
You must configure each Trunk Circuit Routing Group with a Forced Routing Number:
1 2 3 4 5 6
Advanced Configuration Circuit Configuration Trunk Routing Configuration Trunk Circuit Routing Groups Trunk Circuit Routing Group-line Enter a value in the Forced Routing Number field on the Advanced tab.
Recommended to enter XX from DN Channel Map (The Sphericall MGC will use default routing). Note that only numeric entries are allowed. The Quintum gateway copies what you enter here into the username portion of the INVITE "To" header. If you want all calls to go to a specified extension, for example the auto attendant, enter that extension here. If you want to use the default routing defined in the Sphericall Administration utility, you must enter a nonexistent extension here. A nonexistent extension is required since if a valid extension is specified, the MGC will forward inbound calls to it. If a nonexistent extension is specified, the MGC will apply the default routing specified in the Sphericall Administration utility.
Advanced Configuration VoIP Configuration DN Channel Map Click Add Select the channel that corresponds to the FXS port you wish calls to go to
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A local registration may also be created by creating a static SIP route in the var_config file. This method allows the forced routing number to be associated with a SIP phone, but not an FXS port. The var_config.cfg file is a text file that is loaded onto the Quintum gateway via FTP.
Advanced Configuration VoIP Configuration DN Channel Map Click Add Select the channel that corresponds to the FXO port you wish calls to go to
Enter the forced routing number in the DN edit control Uncheck the Register DN check box
A local registration may also be created by creating a static SIP route in the var_config file. This method allows the forced routing number to be associated with a SIP phone, but not an FXO port. The var_config.cfg file is a text file that is loaded onto the Quintum gateway via FTP.
To create a static route, add the following two lines to your existing var_config.cfg file (if any):
1 2
Where <x>=1-5, <forced routing number>=forced routing number and <address> is an IP address or host name.
Open notepad (or any other text editor). Enter the above lines. Save the file as var_config.cfg.
Connect to the gateway via FTP Put the var_config.cfg file into the /cfg folder. After loading the file, the gateway must be rebooted to put the changes into effect. Note: Note that this method has been verified to work with Polycom SIP phones. It
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ports, the From header contains the telephone number of the calling party from the PSTN.
Advanced Configuration VoIP Configuration SIP Signaling Groups SIP Signaling Group-1 User Agent Tab Select the trunk UA and click the Edit button. Enter a password in the Primary Password field
In the Trunk Properties dialog, Authorization tab of the Sphericall Administrator application:
a. Check the Use Authorization check box b. Enter the user name into the Account field (this must be the same as the Primary User
...........................................................
The following line must be added to the var_config.cfg file:
SecureSSG 0 The var_config.cfg file is a text file that is loaded onto the Quintum gateway via FTP. If you already have a var_config.cfg file, simply add the "SecureSSG 0" line to it. If you do not have a var_config.cfg file, open notepad (or any other text editor) and enter the "SecureSSG 0" line.
2
Connect to the gateway via FTP and put the var_config.cfg file into the /cfg folder. After loading the file, the gateway must be rebooted to put the changes into effect.
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...........................................................
Session Initiation Protocol, SIP, is a protocol for transporting call setup, routing, authentication and other feature messages to endpoints within the IP domain. Within the Sphere system, SIP is used to allow external systems to participate in calls with the Sphericall Manager. The Manager targets the use of SIP for integration with some specific third-party products for integration with the following: two-way calls (Sphericall Manager and external system), calls between two systems placed on hold, transfer of calls between the two systems, passing of DTMF digits into the thirdparty system during a call, and notification of message waiting. The Terminal location logic to accommodate gateways that register multiple trunks from the same IP address. Understanding this logic will help enormously when troubleshooting SIP connection issues. The Sphericall Manager (MGC) uses the following logic to locate a SIP terminal in 6.0. and later: From Header Userinfo is compared to the Account field of the Service Provider information. If no match is found, the Sphericall Manager moves to step 2. Note: this is the most common way stations are identified, but does not help for trunks since the FROM field contains the caller ID of the incoming call. To Header Userinfo is compared to the DID maps configured for SIP trunks. If two trunks have overlapping DID maps, the Sphericall Manager moves to step 4. If no match is found, the Sphericall Manager moves to step 3. Contact Header hostname:port is compared to the Outbound Proxy if configured, otherwise the Service Provider Domain of the Service provider information. If the hostname:port is an IP address it is compared exactly to what is configured. If the hostname:port is not an IP address, a partial compare is performed against the Service Provider information. For example, the hostname "horatio.rndlab.spherecom.com" would match the Service Provider information "rndlab.spherecom.com". Note: in both cases the port must match. If more than one User Agent matches this criterion, or no match is found, the Sphericall Manager moves to step 4. Authorization Header If the request contains an Authorization header, the credentials included in the Authorization Header are compared against the credentials configured in the Sphericall Admin Authorization window. If no match is found, the Sphericall Manager moves to step 5.
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Testing
The Sphericall Manager (MGC) challenges the sender to obtain credentials via the Authorization Header.
TESTING
...........................................................
Once installed and configured, Sphere recommends running user tests for connectivity as follows: Dial from one phone to another on the Quintum gateway side. Dial an extension on the main Sphere system from a Quintum side phone. Dial an extension on the Quintum side from the main Sphere system. Dial an call hand transfer it from the Sphere side to the Quintum side. Stop the Sphericall Manager (only during off hours): dial an extension on the Quintum side from the Quintum side. Stop the Sphericall Manager (only during off hours): dial an outbound call from the Quintum side. Stop the Sphericall Manager (only during off hours): dial an inbound auto attendant PSTN number to Quintum side, see if it rings the dedicated failover extension. Test for dial plan. Test for #-codes.
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....
FTP SERVER CONFIGURATION
...........................................................
For the most current information on FTP server installations, see the Microsoft web site or the product documentation for your FTP service.
UPGRADES
Please refer to Sphere Release Notes for any changes to these procedures or upgrades.
CREATE A LOCAL USER ACCOUNT (WITH PASSWORD) FOR THE IP PHONES ON THE FTP SERVER
FTP serverto create login and password for IP phones on FTP server
This account must be created first:
1 2
Create a local user account on the FTP server with username PlcmSpIp OR Sayson.
Create password (respectively; case sensitive)PlcmSpIp OR Aastra480i.
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F T P, S N T P & D H C P N O T E S
3 4 5
Deselect User must change password at next logon. Select User cannot change password. Select Password never expires.
Click Start\Settings\Control Panel\Add or Remove Programs. Click Add or Remove Windows Components.
Select the Application Server. Select the Internet Information Services (IIS) check box. Click Details and ensure that file transfer protocol (FTP) is checked. DISABLE (uncheck) World Wide Web checkbox. Click OK. Click OK. Click Next. Note: You may need to have the Windows Server CD-ROM available when installing
Click Start\Programs\Administrative Tools\Internet Information Services Manager (IIS). Double-click the appropriate server instance in the Tree pane to view all services running on that machine. Right-click Default FTP Site. and Select Properties.
From the Default FTP Site Properties window: In the Identification group box:
16 17 18 19 20
Type a description for the FTP site in the Description field. Select the appropriate IP address for the FTP site from the IP Address drop-down list box. As appropriate for your organizations network and its connection requirements, configure the remaining fields and values in the FTP Site tab. Click Apply. Click the Security Accounts tab.
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22 23
Figure A.1
24
Type or browse to the appropriate path to the FTP site directory in the Local Path field.
The appropriate path to the FTP site directory for a Sphere system is: <hard drive>:\program files\sphere\ftproot. This is the directory in which all XML configuration files will be stored within the Sphere system for use by the IP phones. If you are creating FTP service on the Active Directory or file server, for instance, then the configuration files must be copied to that particular computers directory and the Local Path field must be updated accordingly.
Note: If your organization is using an FTP Server for use with Sphericall IP
phones, and it is not located on the Sphericall Manager, the FTP Server administrator will need to manually copy the FTProot directory to the FTP Server. If you are installing Windows FTP server, the ftproot folder will be located by default at: c:\\inetpubs\ftproot. This default setting needs to be changed as follows: For systems with the FTP Server on the Primary Sphericall Manager, the following folder is required for the location of IP phpone resource files: <drive>:\\Program Files\Sphere\ftproot For systems with the FTP Server on any other server (third-party or Secondary Sphericall Manager), the following folder is required for the location of IP phone resource files: <drive>:\\ftproot\
25 26 27
Select the Read check box. Select the Write check box. Select the Log visits check box.
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28 29 30 31
Click the appropriate radio button, UNIX or MS-DOS, to determine how the FTP server is to present its directories. Click Apply. Click OK. Exit Internet Information Services Manager.
Locate the previously copied FTProot directory on the FTP server. Right click the FTProot directory. Select properties. Select Security tab. Click Add button. Type PlcmSpIp or Sayson (as appropriate). Click Check Names. Click OK. Highlight account. In permissions area: select Full Control. Click OK.
Start an FTP client on a remote machine. Connect to the FTP service running on the Sphericall Manager.
(PlcmSpIp).
If the FTP service has been configured appropriately, a successful login message appears along with the ftp> command prompt.
3 4 5
Type ls to view a listing of the remote directory. Type quit to quit the cmd.exe application. Type exit to exit cmd.exe application.
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F T P, S N T P & D H C P N O T E S
DHCP Configuration
DHCP CONFIGURATION
...........................................................
For the most current information on DHCP server installations, see the Microsoft web site or the product documentation for your DHCP service. IP phones require the following scope options:
Table A.1 Scope Options
Number
002 003 004 066
Action
Time Offset Router Time Server Boot Server Host Name
Notes
See end of DHCP section for table
CONFIGURE DHCP
If IP addresses are to be dynamically assigned throughout the network (and, later, throughout the Sphericall system), a DHCP must be running on at least one server per subnet in all subnets to be configured for DHCP. Refer to the System Requirements documentation for more information concerning DHCP requirements on the network.
Click Start\Settings\Control Panel. Double-click Add or Remove Programs. Click Add or Remove Windows Components to initiate the Windows Components Wizard.
Select the Dynamic Host Configuration Protocol (DHCP) check box. Click OK.
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DHCP Configuration
3 4
Figure A.2
5 6 7
Enter the Name of the scope. Enter the Description of the scope. Click Next.
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F T P, S N T P & D H C P N O T E S
DHCP Configuration
Figure A.3
8 9 10 11
Enter the Start IP address in the open field. Enter the End IP address in the open field. Enter the Subnet mask in the open field. Click Next.
Enter Start IP address of addresses you want to exclude. Enter End IP address of addresses you want to exclude. Click Next.
Enter the Lease Duration Days. Enter the Lease Duration Hours. Enter the Lease Duration Minutes. Click Next.
Enter any Routers or Default Gateways. Click Add to add them to the lower window. Repeat for all Router IP addresses and Gateway IP addresses. Click Next.
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DHCP Configuration
26
Click Next.
Click Yes to activate this score now. Click Next. Click Finish. Click on Action\Authorize.
This action will change Red indication to Green when authorization is established.
33
Click Finish.
To configure the Microsoft Windows 2003 Server DHCP scope for IP Phones only
From the Windows taskbar:
1 2
Click Start\Programs\Administrative Tools\DHCP. Highlight the appropriate scope in the DHCP Tree.
Figure A.4
DHCP window
3 4
Double-click Scope Options in the Contents of Scope pane. Click Action\Configure Options.
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DHCP Configuration
Figure A.5
5 6 7
Add 003 option for Router. Enter IP Address of router. Select the 066 Boot Server Host Name check box.
Adding the 066 Boot Server Host Name option will allow the DHCP server to pass the FTP server address to the IP phone.
8 9 10
Type the IP address of the FTP server in the String Value field. Click Apply. Select the 004 Time Server check box.
Adding the 004 Time Server option will allow the DHCP server to pass the SNTP server address to the IP phone.
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DHCP Configuration
Figure A.6
11 12 13 14
Type the name of the SNTP server in the Server name field. Type the IP address of the SNTP server in the IP address field. Click Apply. Select the 002 Time Offset check box.
Adding the 002 Time Offset option will allow the DHCP server to pass the GMT offset to the phone.
Figure A.7 Scope Options window
15
Type the appropriate unsigned integer as the time offset value in the Long field.
The time offset value is the number of seconds difference from Greenwich Mean Time (GMT), a signed integer. To calculate the time offset value:
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DHCP Configuration
Calculate the number of seconds of offset for the location of your organizations Sphericall system. For example, the GMT offset for Chicago, Illinois (Stardard Time), is -6 hours. -6 hours * 60 minutes/hour * 60 seconds/minute = -21600 seconds
Note: If the value is negative, enter the (-) sign before converting the offset to a
hexadecimal value. Convert the number of seconds of offset to a hexadecimal value. If using a scientific calculator, make certain that the Dword value is selected when converting the hexadecimal value back to a decimal value. -21600 as a hexadecimal value = FFFFABA0 Convert the number of seconds of offset back to a decimal value. The resulting value, in seconds, is the unsigned integer. FFFFABA0 as a decimal value = 4294945696
Note: This calculation is based upon Standard Time. The IP phone automatically
Unsigned Integer
Seattle, WA San Francisco, CA Los Angeles, CA San Diego, CA Calgary, Canada Salt Lake City, UT Albuquerque, NM Boise, ID Chicago, IL New Orleans, LA Houston, TX Mexico City, Mexico New York, NY Toronto, Canada Washington D.C. Miami, FL
Mountain
-7
4294942096
Central
-6
4294945696
Eastern
-5
4294949296
16 17
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DHCP Configuration
Figure A.8
DHCP window
18
A-308
I NDEX
...................................
....
Numerics
480i time server 2-26 time stamp 2-26 480i IP address assignment 2-26 480i IP Phone 2-25
E
Emergency Service Installations 1-2 Erlang model 9-146 Blocking 9-147 Busy hour call multiplier 9-147 Busy hour traffic 9-146 Estimating busy hour traffic 9-148 IVR port 9-147 Recall factor 9-146 Error messages 2-15 Eutectics IPP520 5-109 Eutectics IPP520 Phone configuration 5-109 functionality 5-110 installation 5-109 overview 5-109 Expansion slot types 9-142 Export Phone Distribution Map 3-86 Extensions assigning to a music-on-hold station 7-128 assigning to stations 10-161 configuring for voice mail 10-158
A
Aastra SIP 9112i 2-7, 2-22 SIP 9133i 2-7, 2-22 Aastra 480i IP Phone Installation Static IP Addressing 2-25, 2-26 Aastra 480i login and password 2-7, 222
Call records 10-169 caller ID 3-99, 17-276, 18-295 Caller ID name 18-288 CallNOW implemetation 1-2 CallStateConstants 16-227 Carrier traffic studies 9-148 CBeyond 17-252 Changing the IP Phone User Account 28, 2-23
Aastra 9133i SIP Phone 3-89 Aastra password 2-23 Aastra SIP Phone 3-52 account on the domain 2-8, A-297 A-law 18-289 AudioCodes access and setup 14-203,
14-212
B
Bogen Transformer 7-124 Boot Screen Messages 2-15 Boot Sequence Errors 2-15 BranchHub Manual 1-3 Busy Lamp Field 2-29 Busy Lamp Field States 2-29 Busy-Lamp-Field 2-28
Changing the SNMP write community 12-186 COHub Manual 1-3 COM API Getting Started 16-228 Com API 16-228 Common message coding 10-170 blocked message waiting 10-171 direct call message 10-170 forward all calls 10-170 forward on no answer 10-170 forward on unknown reason 10-171 invalid message waiting 10-171 message waiting indicator messages 10-171 configure a voice mail station 10-159 configure the VM and AA 10-162 configure the voice mail line setting 10161
F
Fax Relay 18-289 File Upgrade 2-17 Flow control settings 10-154 FP *96 9-150, 10-161 FTP Preparation 2-9 FTP server 2-13, A-297 configuring A-297 installation A-298 moving the service A-297 upgrades A-297 FTP server address 2-6, 2-20 FTP Server Configuration Topics A-297 FTP Service A-298 FTP service functionality A-300
Constants 16-227
D
Daily Management 1-2 DHCP 2-25, A-301 configure service A-304 install service A-301 DHCP Configuration A-300, A-301 DHCP scope for IP Phones A-304 Diagnostic Star Codes 3-98 DID maps SIP 3-99, 17-276, 18-295 DNS domain name 17-243 DNS Test 3-84 DNS test 17-244 Drop loop current 7-128, 8-136 Dynamic Host Configuration Protocol (DHCP) A-301
C
cable 2-25 Call loggers 9-147 Call logging 7-128, 8-136, 10-160 Call progress tones 9-149 amplitude levels 9-150 busy 9-149 disconnect 9-149 external dial tone 9-149 flashtime 9-150 internal dial tone 9-149 reorder 9-149 ringback 9-149
G
G711 caveat for Global Crossing 17-241 Generic SIP Trunk 17-244 Getting Started Sphericall COM Type Libraries 16228
Index
I-1
INDEX
Grandstream GXP-2000 speed dial configuration 3-63, 3-71 time zone configuration 3-63, 3-71 upgrades 3-64, 3-72 Grandstream GXP-2000 Phone 3-58, 366
media servers creating 10-164 MeetingHub Manual 1-3 Memory 9-142 Message Server voice port requirements 9-146 Message Storage Retrieval Identifier 9144
O
OID index 12-191 OIDs 12-186 Outbound Proxy 3-99, 17-276, 18-295
H
Hangup detection 9-150 Hard disk storage 9-142 Hardware Manuals BranchHub 1-3 COHub 1-3 MeetingHub 1-3 MG Command Line Reference 1-3 PhoneHub 1-3 Help for programming ideas 16-227 Help with the Visual Basic 16-227 hostname 3-99, 17-276, 18-295 hunt order auto attendant extension 10-162 voice mail extension 10-162
Message stores MSRLineID 10-165 MG CLI Reference Manual 1-3 MGC-to-IP phone connections 2-12, 224
P
Paging lines 8-133 adding a line 8-134 configuration 8-134 installation and integration 8-136 profile 8-136 Paging system installation test 8-138 recommended products 8-133 zone configurations 8-136 password 2-8, A-297 PBX properties configuring 2-11 PBX statistics 9-147 Phone failover 2-19 Phone Setup 2-25 PhoneHub Manual 1-3 phones 5-109 Aastra 480i 2-20 PING 2-16 Plan & Prepare the Sphere System 1-2 Planning 2-20 Plantronics 5-112 CS50-USB 5-112 platform hardware requirements 9-142 Polycom IP phone account permissions 2-7, 2-22 Polycom IP501 Polycom IP601
I
Install & Configure the Sphere System 1-2 Install Sphericall Voice Mail 1-2 Integrate Partner Technologies 1-2 Integration notes 9-149 IP address 2-6, 2-20, 3-99, 17-276, 18295
MGC-to-MG Connection Control configuring 2-12, 2-24 MGCP phones 2-12, 2-24 MIBs installation on the Sphericall Manager 12-189 installing 12-189 sphere-reg.mib 12-189 sphere-tc.mib 12-189 table of channel description objects 12-190 table of e2prom objects 12-190 table of system objects 12-190 table of telephony board description objects 12-190 table of version objects 12-190 table of voltage/temperature objects 12-191 Microsoft Windows Server 2-6, 2-20 MLPP 1-2 MOH and Upgrades 7-130 MOH behavior during MGC timeout 7130
IP Phone previously on network 2-9 IP phone initialization 2-9 IP Phone Installation Dynamic IP Addressing 2-12 IP Phone Upgrades 2-19, 2-39 IVR port 9-147
L
Link Wireless Telephone 15-223 Load Balancing for Sayson IP Phone 211, 2-24
local user account 2-23 Logins & Permissions Group Policy for Windows Installer 26, 2-21
MoH for fallback 7-125 MoH Options 7-125 MOH Upgrade path 7-129, 7-130 Moves, Adds & Changes to the System 1-2 MSRLineID 9-144, 10-165 Mu-law 18-289 Music-on-Hold 7-123 Bogen Transformer 7-124 device sources 7-124 interface device 7-124 moh1 7-128 music source 7-123 re-broadcasting rights 7-124 MWI light flashes 10-168
3-
Primary addresses group extensions and voice mail extensions 7-129 Processor 9-142
Q
Quick Reference Guides 1-2 quick start guide 2-12 Quintum 18-279 Caller ID on the station ports 18-288 Hook Flash 18-288
N
Network DHCP A-301 Normal Boot Screens 2-15 normal messages during boot sequence 2-15 Numbering plan
M
MAC address 17-246 Manage, Monitor & Support Sphericall 1-2 Media servers 10-164
R
register SIP 3-98, 17-276, 18-295 Registration Type 17-238 Release Notes & Upgrade
I-2
Index
INDEX
Procedures 1-2 report phone distribution map 3-86 Reports, Statistics and Tools 1-2 Reset 2-33 Restarting the Ip Phone 2-16 Restarting the phone 2-16 RS-232 serial connection 9-139, 9-141
272
S
Sample Applications for Sphericall Type Libraries 16-228 SCCall 16-227 SCCalls 16-227 scope DHCP Scope Range A-304 Scope Options A-301 SCPhone 16-227 Screen Messages 2-15 Service Provider 3-99, 17-276, 18-295 Service Provider Domain 3-99, 17-246,
17-276, 18-295
softtrunks 17-238 System Initialization Settings 17-242 Tie line solution 17-253 Tie-lines 17-239 Unknown authentication 17-249 User Agents 3-45, 17-234, 17-254 User Agents Upgrade 3-48, 17-237 User Name/User ID 3-42 Userinfo 3-98, 17-276, 18-295 SIP Configuration Note 3-42 SIP connection overview 3-98, 17-276,
18-295
SIP Failover 3-49 SIP phone restart 3-65, 3-73 upgrade 3-65, 3-73 SIP phones restart 3-57 SIP phones upgrade firmware 3-57 SIP Registration 17-238 SIP Server 3-42 SIP Terminal location logic 3-98, 17-276,
18-295
Account 17-241, 17-246 Account field 3-98, 17-276, 18-295 Authentication 3-42 Authorization 17-248 Call center 17-239 Challenge authentication 17-249 Contact Domain 17-243 credentials 3-99, 17-277, 18-296 Description of trunk 17-241 DNS Name 3-43 Inbound Registration 17-239 INVITE 17-249 IP Address 3-43 Outbound Proxy 3-43, 17-242 Outbound Registration 17-239 Planning 17-240 Port 17-246 Port information 17-242 Primary MGC 17-243 Proxy 3-43 REGISTER 17-249 REGISTER requests 17-243 Registrar 3-42 Registration Type 17-246 Registration type 17-242 Requirement 3-45, 17-234, 17-254 Respond authentication 17-249 Secondary MGC 17-243 Service Provider 17-239 service provider 17-246 Service Provider Domain 17-242 SIP Domain 3-43 SIP Phone MAC Address 3-42 Softtrunk Total Capacity 17-247, 17-
SIP tie line to Call center 17-266 SIP trunk configuration Global Crossing Requirements for SIP 17-241 SIP Trunk MAC Address 17-244 SIP Trunking Configuration 17-252 SIP Trunking Tie Line 17-253 SIP Trunking to SIP Service Provider 17-239 SIP trunks DID maps 3-99, 17-276, 18-295 SMDI 9-139, 10-157, 11-173 access hangup detection 9-150 Auto Attendant Group Number 9-144 cables 9-141 cabling requirements 9-141 Call Center 9-139 call processing 9-144 call records 10-169 common message coding 10-170 enabling SMDI process 10-156 Fax Center 9-139 hardware requirements 9-140 integration notes 9-149 Message Storage Retrieval Identifier 9-144 messages from the voice mail system 9-140 messages to the voice mail system 9140
RS-232 serial connection 9-139 setting flow control settings 10-154 setup and operation 9-143 supported features 9-140 system planning 9-146 transfer initiation and release 9-150 Voice Messaging 9-139 voice messaging platform requirements 9-141 SMDI integration 9-139 SMDI platform hardware requirements 9-142 SMDI process 10-156 SNMP agent 12-186 definition 12-185 manager 12-185 manager and agent setup 12-187 monitoring process 12-187 OID index 12-191 OIDs 12-186 overview 12-185 requirements 12-186 Sphericall MIBs 12-189 Sphericall traps 12-188 trap function 12-187 trap information 12-188 traps 12-186 SNMP access 12-186 SNMP Agent 12-186 SNTP server 2-13 SNTP server address 2-6, 2-20 SNTP Server Configuration A-300 SNTP time offset 2-13 softtrunk 17-238 SoundPoint MGCP Phone Installation Static IP Addressing 2-13 Spanish 1-2 Spectralink 15-223 Sphere preparing the system 10-154 Sphere Document Index 1-2 Sphere Star Codes 1-2 Sphere System Requirements 1-2, 2-6,
2-20
MSRLineID 9-144 overview 9-139 requirements 9-140 RS-232 compliant serial connector 9141
Sphericall COM API 16-228 Sphericall Desktop Users Manual 1-2 Sphericall Manager Primary vs Secondary 10-156 Sphericall Manager Commissioning wizard choosing a Pirmary or Secondary Sphericall Manager 10-156 choosing voice mail 10-157 Sphericall Manager Configuration utility SMDI 10-157 Sphericall Phone Type Library 16-227 Star Codes administrative 3-97 static IP address 2-13, 2-26
Index
.....
I-3
INDEX
Static IP Addressing Aastra 480i phones 2-25, 2-26 Polycom SoundPoint phones 2-13 Station settings call logging 7-128, 8-136, 10-160 drop loop current 7-128, 8-136 extension assignment 10-161 properties 8-135, 10-160 stutter dial tone 7-128, 8-136 Stutter dial tone 7-128, 8-136 Support Diagnostic Star Codes 3-98 Symptoms 11-181 Sync IP Phone Files 2-16
user settings 3-93 web configuration 3-91 WiFi and network settings 3-90 wireless access point settings 3-93 UTStarcom F1000G overview 3-89 UTStarcom F3000 overview 3-89
V
verify Monitor privileges for the SMDI instance 10-163 view all the SNMP community names 12-187 VLAN use for IP phone 2-9 Voice 11-180, 11-181 voice mail extensions configuring 10-158 voice message deletion 10-168 voice message storage 9-142 Voice messaging configuring 10-164 media servers 10-164 Message Waiting Indicators 10-169 platform requirements 9-141 restarts and refreshes 10-169 system testing 10-166 voice part requirements 9-146 volume 18-288 volume settings 5-108
T
T.38 Fax Relay 18-289 test coverage to voice mail and to test call disconnect 10-168 test direct calls into voice mail by the operator 10-167 test direct subscriber access of mailboxes 10-167 test the setting and cancelling of MWI 10-167 Third-party permissions required 2-6, 221
Time offset values A-307 To change the SNMP access write community name 12-186 Traps 12-186 critical information 12-188 function 12-187 Sphericall traps 12-188 Troubleshooting 2-30 error messages 2-15 phone failover 2-19 restarting the phone 2-16 troubleshooting SIP 3-98, 17-276, 18-295 SIP stations 3-99, 17-276, 18-295 SIP trunks 3-99, 17-276, 18-295 Troubleshooting information HyperTerminal monitoring 11-173,
11-177
W
Web Client 2-26
X
XML file 2-23
U
Upgrades 2-19, 2-39 user account 2-8, 2-23, A-297 User Agent 3-99, 17-276, 18-295 user agent 3-46, 17-235 User Agents 18-286 UTStarcom 2-8, 2-22 UTStarcom F1000 create a DNS Record 3-49, 3-50 firmware updates 3-94 Sphericall Voice Mail settings 3-94
I-4
Index
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