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Voice Gateways System Manual

1. HTTP LAN Yes No Yes checked HTTP WAN Yes No Yes checked SNMP LAN Yes No Yes checked SNMP WAN Yes No Yes checked Upgrade Page Upgrade Type Auto TFTP HTTP TFTP Host URL IP address Domain Name Null File name A string of up to 25 characters 78 Operation A Appendix A Internal Class 5 Services In This Appendix This appendix provides a description of the internal Class 5 services that are supported by the Voice Gateway Appendix A Internal Class 5 Services A 1 Actions and Keypad Sequences Table A 1 Keypad Sequences Action Description Keypad Sequence R hook flash HOLD Holds an on going call RO DROP Drops an on going call R1 FLASH Switches between on going call R2 sessions or starts new call inquiry CONFERENCE Activates 3 party conference R3 CONFERENCE DROP Deactivates 3 party conference R5 CW ACTIVATION Enables Call Waiting indication 43 tone CW DEACTIVATION Disable Call Waiting indication 43 tone CW STATUS CHECK Informs about the current 43 configuration of Call Waiting indication tone CALL FORWARD ACTIVATION Call Forward activation 21 lt telephone number gt CALL FORWARD Call Forward de activation 21 DEACTIVATION 80 Voice Gateways System Manual Using the Class 5 Services A 2 Using the Class 5 Services A 2 1 Call Waiting Description One on going call active audible CW tone indicatin
2. Enable DRAP Enable Disable Disable unchecked Enable Pre allocation Enable Disable Enable checked DRAP Server IP Address IP address or null for auto Null auto discovery discovery Server Port 8000 8200 8171 Discovery Time 1 255 seconds 10 seconds Acknowledge Time 1 to 10 x 100 milliseconds 3 300 milliseconds Clear Count 1 10 2 Retry Count 1 10 5 RTP Packing Ratio 10 to 100 milliseconds in 30 milliseconds multiples of 10 10 20 100 Vocoder Type G711 G729 G729 Voice Gateways System Manual 77 Chapter 3 Using the Web Configuration Server Table 3 18 Parameters Summary Parameter Range Options Default Set security Password Page New password Confirm new password Up to 20 printable characters case sensitive A null empty string means no password Localization Page NTP Server IP address or null for disable Null NTP server disabled NTP server Time Zone Drop Down Menu GMT 01 00 Adjust clock for daylight savings Yes No Yes checked SNMP Configuration Page Trap Destination 1 to Trap destination 6 IP addresses If all Trap Destinations are null SNMP traps will be sent as Null for all addresses Service Access Configuration Page broadcasts Read Community Up to 20 printable characters public case sensitive Write Community Up to 20 printable characters private case sensitive
3. Table 3 3 WAN Configuration Page Parameters Parameter Description Client identity Applicable only if the Obtain WAN configuration dynamically option is selected The method used for identifying the client Option 61 The options are Standard The unit s MAC address Custom An identification string of up to 25 characters The default is null an empty string Vendor ID Applicable only if the Obtain WAN configuration dynamically option is selected The Vendor ID Option 60 A string of up to 25 characters The default used by the unit is VoIP not displayed Specify static WAN configuration Select this option to configure the IP parameters manually See also Section 2 2 2 IP Address Applicable only if the Specify fixed WAN configuration option is selected The IP address of the unit The default is 192 168 254 254 Subnet Mask Default Gateway Applicable only if the Specify fixed WAN configuration option is selected The IP Subnet Mask The default is 255 255 255 0 Applicable only if the Specify fixed WAN configuration option is selected The Default Gateway address The default is none empty DNS Address Applicable only if the Specify fixed WAN configuration option is selected IP DNS Server address The default is none empty Host Name The Host name for clients A string of up to 25 characters The default is null an empty string Domain Na
4. Chapter 1 System Description 1 2 3 Voice Quality Table 1 3 Voice Quality Item Description Voice Codecs E G711 Ulaw E G711 Alaw E G 729ab Prioritization E IEEE 802 1p layer 2 prioritization E DiffServ layer 3 prioritization General E Adaptive jitter buffer E Echo cancellation E Speech sampling rate 10 60 ms E Silence suppression with comfort noise 1 2 4 Configuration and Management Table 1 4 Configuration and Management Item Description Management Options E Internal Web Server E SNMP SNMP Agents SNMPV1 clientMIB Il RFC 1213 Private MIB Plug amp Play Functionality DHCP including support messages option 60 61 43 Software Upgrade Using TFTP Configuration Download Using TFTP 1 2 5 Bridge Functionality Table 1 5 Bridge Functionality Item Description Supported Ethernet Devices Up to 32 MAC addresses 4 System Description Specifications Table 1 5 Bridge Functionality Item Description Unknown address Forwarding Forward Unknown Policy Bridge Aging Time 180 seconds 1 2 6 Mechanical Table 1 6 Mechanical Specifications Item Details Dimensions W x D x H 17 6 x 11 x 2 8 cm Weight 230g 1 2 7 Electrical Table 1 7 Electrical Specifications item Details Power Input 12 VDC fro
5. SIP SIP Extensions NAT STUN Client ToS Line Configuration Home VLAN ToS Decimal Telephone Call Signaling Packets fo RTP Packets o 3 SNMP Packets jo Logout Default setting jo Save ToS Settings Figure 3 18 ToS Page The ToS page includes the following components Table 3 12 ToS Page Parameters Call signaling Packets ToS marking for call signaling packets Enter a number in the range 0 to 255 The first 6 bits is the value of the DSCP field or null The default is 0 RTP Packets ToS marking for RTP and RTCP packets Enter a number in the range 0 to 255 The first 6 bits is the value of the DSCP field or null The default is 0 SNMP Packets ToS marking for SNMP packets Enter a number in the range 0 to 255 The first 6 bits is the value of the DSCP field or null The default is 0 Default setting ToS marking for other types of packets e g HTTP TFTP Enter a number in the range 0 to 255 The first 6 bits is the value of the DSCP field or null The default is 0 Click on the Save ToS Settings button before leaving the page to save the new settings The new settings will be applied after restarting the unit 3 7 6 Line Configuration Page The Line Configuration page enables to select the country standard for Caller ID When using a caller ID device select your country standard from the list Voice Gateways System Manual 53 Chapter 3 Using the
6. StringList DialPlan String StringList 0 9 denotes a single digit between O and 9 To configure a range of more than 10 numbers e g 800xxx 8 19xxx use the scheme 80xxxx 81 0 9 xxx A dialplan according to this syntax is defined either by a case insensitive string or by a list of strings Regardless of the above syntax a timer is only allowed if it appears in the last position in a string 12T3 is not valid Each string is an alternate numbering scheme The unit will process the dialplan by comparing the current dial string against the dialplan If the result is under qualified partial matches at least one entry then it will do nothing further but wait until a full match is reached If the result is over qualified no further digits could possibly produce a match then it aborts the dial attempt and notifies end user with an audio signal Only a full match will trigger to initiate a call by sending the dialed information to a Gatekeeper SIP server The Timer T is activated when it is all that is required to produce a match The period of timer T is 4 seconds as default configurable For example a dialplan of xxxT xxxxx will match immediately if any 5 digits are entered It will also match following a 4 second pause after entering 3 digits IMPORTANT The dialplan is according to section 2 1 5 of RFC 3435 The notation denotes zero or more keys That is x means none or at least one digit followed
7. on page 3 17 Using the Web Configuration Server on page 3 18 Home Menu Product info Page on page 3 20 WAN Menu on page 3 22 VLAN Tagging Menu on page 3 26 Telephone Menu on page 3 38 BW Reservation DRAP Configuration Page on page 3 58 System Menu on page 3 62 Upgrade Page on page 3 68 Restart Page on page 3 71 Logout Page on page 3 72 Parameters Summary on page 3 73 Chapter 3 Using the Web Configuration Server 3 1 Introduction to the Web Configuration Server The Voice Gateway can be configured using the following methods E The Web Configuration Server E An ini file loaded into the unit from a TFTP server or automatically downloaded using DHCP option 43 This document describes the configuration using the Web Configuration Server 16 Operation Accessing the Web Configuration Server 3 2 Accessing the Web Configuration Server To manage the unit you must have prior knowledge of its WAN IP Address Follow the steps below to access the Web Configuration Server 1 Open a web browser 2 Enter the WAN IP address of the unit in the Address field of the browser and click Enter E g http 192 168 254 254 default 3 Ifthe Web Configuration Server is password protected you will be prompted to enter your user name and password in order to login to the system To login with operator privileges full access and read write
8. Call Waiting Tone The tone that the end user that originates the call hears when there is a second incoming call during the original call session Cadence 0 2 second on 0 5 second off 0 2 seconds on Duration One On Off On cycle Frequency 425 Hz Level 10 dBmO 86 Voice Gateways System Manual Table B 1 Default Telephony Parameters Parameter Definition Default Caller ID Standard The country standard and data transmission method Sweden DTMF CID Alerting Method The method of alerting on the existence of CID data Polarity Reversal On hook data transmission timing method The timing for transmission of CID data Post first ring signal 2 Wire Impedance The impedance presented between the A and B wires of the telephone line in active state nominal impedance CTR21 ETSI complex 2700 750Q 150nF Voice Gateways System Manual 87 AS NZS BW CLIP CNI CSA CW DHCP DiffServ DNS DRAP DSCP DTMF EC Glossary Australian and New Zealand Standard Band Width Calling Line Identification Presentation A supplementary service used to show the number of a caller Calling Number Identification Canadian Standards Association Call Waiting Dynamic Host Configuration Protocol A protocol for dynamically assigning IP addresses from a pre defined list to nodes on a network Using DHCP to manage
9. Networking Gateways option of the Configuration menu for a selected SU The displayed information includes Gateway s type IP Address and the VLAN ID used for management In general the same VLAN should be configured in the Voice Gateway for Management Default VLAN ID and Voice RTP and Signaling as the Voice Gateway uses one IP address for two VLANs and the default router in the backbone cannot operate in this mode To support the required quality of service when DRAP is used provision the correct VoIP Service If DRAP is not used provision an L2 Service with a CG connection refer to the BreezeMAX System Manual for details 2 4 2 BreezeACCESS VL System Version 3 1 To access the Monitor program of the SU from a PC connected to the LAN port of the Gateway the WAN port must be configured with static IP address that is in the same subnet as the IP Address of the SU and subnet mask 255 255 255 0 the default IP address is 10 0 0 1 with a Subnet Mask 255 255 255 0 The PC used for accessing the Monitor program should be configured to belong to the same subnet Configure the Traffic Prioritization parameters in both the SU and the AU to ensure high priority for RTP traffic Refer to the BreezeACCESS VL System Manual for details Installation Chapter 3 Using the Web Configuration Server In This Chapter Introduction to the Web Configuration Server on page 3 16 Accessing the Web Configuration Server
10. is SS disabled EC The EC Echo Cancellation option defines whether to activate the echo cancellation mechanism for improved voice quality EC is not used during Fax T 38 transmissions The default is enabled Packet The packet size in milliseconds The range is from 10 to 150 milliseconds The default is 30 ms for G729 and 20 ms for G711A and G711U Keypad The Keypad field indicated which transmission method to be used for user input DTMF signaling i e phone banking None means in band which should be used with G 711 only For SIP model the options are None RFC2833 and SIP INFO RFC2833 and SIP INFO should be used primarily with G 729 but could also be used with G 711 The default is None for G711 codecs and RFC2833 for G729 For H323 model the options are H225 H245 RFC2833 and None The default is None for G711 codecs and H225 for G729 Priority The Priority parameter defines the relative priorities to be offered during capabilities exchange If only G711A and G711U are used the permitted priorities are 1 and 2 If all 3 codecs are used the permitted priorities are 1 2 and 3 Voice codec negotiation priority is always performed between 2 end points and depending on which side initiated the negotiation The default is Priority 1 to G711A Priority 2 to G711U Voice Gateways System Manual 47 Chapter 3 Using the Web Configuration Server Table 3 8 Codecs and Fax Configur
11. protocol that can establish modify and terminate multimedia sessions such as Internet telephony calls VoIP SIP can also invite participants to already existing sessions such as multicast conferences Media can be added to and removed from an existing session SIP transparently supports name mapping and redirection services which supports personal mobility users can maintain a single externally visible identifier regardless of their network location Voice Gateways System Manual Glossary SNMP ss STUN SU T 38 TCP TCP IP TFTP ToS UDP Simple Network Management Protocol A network management protocol that provides a means to monitor and control network devices and to manage configurations statistics collection performance and security SNMP works by sending messages called protocol data units PDUs to different parts ofa network SNMP compliant devices called agents store data about themselves in Management Information Bases MIBs and return this data to the SNMP requesters Silence Suppression A method for eliminating wasted bandwidth when sending voice over a packet switched system Simple Traversal of UDP A method of NAT traversal through UDP based on RFC 3489 Subscriber Unit This is the ITU T recommendation which defines a real time method for fax over IP networks Transmission Control Protocol Connection oriented transport layer protocol that provides reliable full duplex d
12. the primary Gate Keeper provides to the Voice Gateway during registration an IP address for the secondary system Gate Keeper IP secondary H323 model only The IP address of the secondary Gate Keeper 42 Operation Telephone Menu Table 3 6 SIP Configuration H323 Telephone Page Parameters Parameter Description User Name SIP model only The SIP user Name Format name or number depends on the SIP server A string of up to 25 characters Password SIP model only H323 Alias The SIP user Password Format name or number depends on the SIP server A string of up to 25 characters The unit s name used when registering the unit at the Gate Keeper If used the H323 alias must be set to a unique value for each telephone line in the network in order for the system to accept it Up to 25 characters The default is null not used during registration Outgoing Display name The name to be displayed on the caller ID display of a receiving party if supported by the network Up to 25 characters with no spaces Telephone number The telephone number of the specific telephone line to be used when registering the unit at the Gate keeper SIP Server The telephone number is limited to 25 characters It may also be an e mail address limited to 25 characters before the sign The Telephone number must be set to a unique value for each telephone line in the n
13. 40 milliseconds Automatically switch to Fixed Jitter Buffer Select this option in order to use both fax and voice The Voice Gateway automatically switches to the configured Fixed Jitter Buffer upon detecting a fax modem tone Faxes can be transmitted when Codec G 711 or T38 are selected The following settings are available for each line Table 3 8 Codecs and Fax Configuration Parameters Parameter Description Codec The Codec check boxes identify which codecs are used By default all three codecs are selected checked NOTE G 729 with Annex A is implemented in the Voice Gateway It enables communication with devices using either G729 with Annex A or G729 with Annex A and Annex B It is not possible to communicate with devices using G729 with Annex B only For each Codec in use the following can be configured 46 Operation Telephone Menu Table 3 8 Codecs and Fax Configuration Parameters Parameter Description Ss The SS Silent Suppression option for outgoing calls When the SS option is enabled silence intervals are identified and only relevant information is transmitted using less bandwidth than during voice activity intervals This allows for a better overall utilization of the available bandwidth It is possible to enable Silent Suppression with G729 codec Silent Suppression is not applicable when using the G711 codecs The default G729
14. Chapter E Installation Requirements on page 2 8 E Front and Rear Panel Components on page 2 9 Installation and Commissioning on page 2 11 Notes on Using the Voice Gateways in Alvarion s Systems on page 2 14 Chapter 2 Installation 2 1 Installation Requirements 2 1 1 Packing List E Voice Gateway with one VG 1D1V or two VG 1D2V Phone Ports E Power supply with a DC connecting cable E Mains power cable 2 1 2 Additional Installation Requirements E A straight Ethernet cable for connecting the WAN port to the SU IDU E An Ethernet cable for connecting to the user s data equipment straight for connecting to a PC crossed for connecting to a hub switch E Standard phone cable s with RJ 11 connectors E Mains plug adapter if the power plug on the supplied mains power cable does not fit local power outlets E Portable PC with an Ethernet card and an Ethernet cable for configuring the Voice Gateway parameters using a web browser 8 Installation Front and Rear Panel Components 2 2 Zad Conne B ctors Front and Rear Panel Components aa 1N DCO Se e RESET WAN Phone 1 Phone 2 NOTE x Figure 2 1 Voice Gateway VG 1D2V Back Panel The VG 1D1V has a single Phone connector Table 2 1 Voice Gateway Connectors Name Connector Functionality Phone 1 RJ 11 Connections to the user s telephones Phone 2 VG 1D2V only RJ 11 Connections to the us
15. IP addresses simplifies client configuration and efficiently utilizes IP addresses See DSCP Domain Name System The name resolution system that lets users locate computers on the Internet TCP IP network by domain name The DNS server maintains a database of domain names host names and their corresponding IP addresses Dynamic Resource Allocation Protocol Differentiated Service Code Point AKA DiffServ An alternate use for the ToS byte in IP packets Six bits of this byte are being reallocated for use as the DSCP field where each DSCP specifies a particular per hop behavior that is applied to the packet Dual Tone Multi Frequency The type of audio signals that are generated when you press the buttons on a touch tone telephone DTMF assigns a specific frequency consisting of two separate tones to each key Echo Cancellation Glossary EMC FCC FSK G 711 G 729 H 225 H 245 H 323 HA 90 Electro Magnetic Compatibility The capability of equipment or systems to be used in their intended environment within designed efficiency levels without causing or receiving degradation due to unintentional EMI Electro Magnetic Interference EMC generally encompasses all of the electromagnetic disciplines Federal Communications Commission A U S government agency that supervises licenses and controls electronic and electromagnetic transmission standards Frequency Shift Keying A simple modulation techniq
16. Priority O for data on the WAN port and untagged to from the LAN port Operation VLAN Tagging Menu VLAN200 VLAN 100 Data Voice amp Management Untagged Figure 3 9 VLAN Configuration Example 1 1 Inthe VLAN page click Add VLAN to open the VLAN Editor 2 Inthe VLAN Editor enter the follwing for Voice and Management VLAN VLAN ID 100 VLAN NAME Voice amp Mng VLAN Priority 5 WAN Yes gt o LAN No 3 Click OK to add the VLAN to the Tagged Port Membership table 4 Enter the VLAN ID for Voice and Management 100 in the field Default VLAN ID on WAN port and click Save 5 In the Page Selection bar click on VoIP VLAN Configuration to open the VoIP VLAN Configuration page Enter 100 in the VLAN Tag fields for both Call Signaling and RTP Enter 7 in the Priority Tag field for both Call Signaling and RTP Click Save VoIP VLAN Settings Go back to the VLAN Tagging page 6 Inthe VLAN page click Add VLAN to open the VLAN Editor to configure the data VLAN Voice Gateways System Manual 31 Chapter 3 Using the Web Configuration Server 7 Inthe VLAN Editor enter the follwing for data VLAN ID 200 an arbitrary selection a VLAN ID is required for defining the untagged data This VLAN tag is only used internally in the unit VLAN NAME Data VLAN Priority O WAN Yes LAN Yes 8 Click OK to add the VLAN to the Tagged Port Membership table 9 Enter the VLAN ID
17. Web Configuration Server 3 7 54 Extensions NAT STUN Client ToS DifSew Line Configuration VLAN Line Configuration Telephone CLIP Standard SWEDEN 7 Upgrade Save Restart Logout Figure 3 19 Line Configuration Page Click on the Save button before leaving the page to save the new settings The new settings will be applied after restarting the unit The default is Sweden Dial Plan Schemes A dialplan gives the unit a map to determine when a complete number has been entered and should be passed to the gatekeeper SIP server for resolution into an IP address Dialplans are expressed using the same syntax as used by MGCP NCS specification The following notation describes the formal syntax of the dialplan Digit O 1 2 3 p 4 5 6 7 p 8 9 Timer T t Letter Digit Timer A a B p b C e D d Range X x matches any digit Letters matches any of the specified letters Letters Subrange Subrange Letters Subrange Letter matches the specified letter Digit Digit matches any digit between first and last Position Letter Range StringElement Position matches any occurrence of the position Position matches an arbitrary number of occurrences including 0 String StringElement StringElement String Operation Telephone Menu StringList String String
18. a separate entity Interactive Voice Response A telephony technology in which someone uses a touch tone telephone to interact with a database to acquire information from or enter data into the database Local area Network A computer network limited to a small geographical area such as a single building The network typically links PCs as well as shared resources such as printers Media Access Control The lower of the two sub layers of the data link layer defined by the IEEE The MAC sub layer handles access to shared media such as whether token passing or contention will be used Standardized data link layer address that is required for every port or device that connects to a LAN Other devices in the network use these addresses to locate specific ports in the network and to create and update routing tables and data structures MAC addresses are 6bytes long and are controlled by the IEEE Voice Gateways System Manual Glossary MDI MDIX MDI MDIX is a type of Ethernet port connection using twisted pair cabling The MDI for Medium Dependent Interface is the component that provides the physical and electrical connection to the cabling medium An MDIX for MDI crossover is a version of MDI that enables connection between like devices MDI ports connect to MDIX ports via straight through twisted pair cabling both MDI to MDI and MDIX to MDIX connections use crossover twisted pair cabling The design of Auto MDI MDIX or
19. by Alvarion Ltd will void equipment warranty and any repair thereafter shall be charged for It could also void the user s authority to operate the equipment Voice Gateways System Manual vii About This Manual This manual describes Alvarion s Voice Gateway units and how to install operate and manage them Version R2H276 supports SIP only This manual is intended for technicians responsible for installing setting up and operating the Voice Gateway and for system administrators responsible for managing the Voice Gateways This manual contains the following chapters and appendices Chapter 1 System Description Describes the Voice Gateway and its functionality Chapter 2 Installation Describes how to install the Voice Gateway and connect it to the SU and to the user s equipment Chapter 3 Using the Web Configuration Server Describes how to use the Web Configuration Server for configuring parameters and checking system status Appendix A Internal Class 5 Services Describes the internal Class 5 services that are supported by the Gateway Appendix B Default Telephony Parameters Describe the default values for some telephony parameters including signals tones parameters CID parameters and line impedance Glossary Provides definitions of various terms used in the manual Contents Chapter 1 System Description 1 1 introducing the Voice Gateway iis ccciccsccctccsssccccercnss cncesencsectnacntceteead
20. want to log out the system Upgrade TERE gt t Logout Figure 3 28 Logout Page 72 Operation Parameters Summary 3 13 Parameters Summary Table 3 18 Parameters Summary Parameter Range Options Default WAN Configuration Page Device Operating Mode Only Bridge option is available Bridge Obtain WAN configuration Yes checked No Default Yes selected dynamically unchecked Factory default above 10 sec HW reset No unchecked Client identity Standard Custom Standard The Custom string can include up to 25 characters The default Custom string is null Vendor ID A string of up to 25 characters VoIP used by default but is not displayed Specify fixed WAN configuration Yes checked No No unchecked unchecked IP Address IP address Default Dynamic Factory default above 10 sec HW reset 192 168 254 254 Subnet Mask IP address 255 255 255 0 Default Gateway IP address Null DNS Address IP address Null Host Name A string of up to 25 characters Null Domain Name A string of up to 25 characters Null VoIP VLAN Configuration Page Call Signaling VLAN Tag 1 4094 or null Null Call Signaling Priority Tag 0 7 or null Null RTP VLAN Tag 1 4094 or null Null RTP Priority Tag 0 7 or null Null SIP Configuration H323 Telephone Page Voice Gateways System Manual Chapter 3 Using the Web Conf
21. widely accepted standards PPP and Ethernet PPPoE is a specification for connecting the users on an Ethernet to the Internet through a common broadband medium such asa single DSL line wireless device or cable modem All the users over the Ethernet share a common connection so the Ethernet principles supporting multiple users in a LAN combines with the principles of PPP which apply to serial connections Ringer Equivalency Number A number determined in accordance with the Code of Federal Regulations Title 47 part 68 which number represents the ringer loading effect on a line A REN of 1 represents the loading effect of a single traditional telephone set ringing circuit Modern telephone instruments may have a REN lower than 1 Relative Humidity The amount of water in the air relative to the maximum amount of water that the air can hold at a given temperature RTP Control Protocol A protocol that monitors the QoS of an RTP connection and conveys information about the on going session Real Time Protocol An Internet protocol for transmitting real time data such as audio and video RTP itself does not guarantee real time delivery of data but it does provide mechanisms for the sending and receiving applications to support streaming data Typically RTP runs on top of the UDP protocol although the specification is general enough to support other transport protocols Session Initiation Protocol An application layer control IETF
22. 1 February Example 2006 3 7 1 SIP H323 Configuration The page was updated support of Version 1 1 February Page Message Waiting 2006 3 7 1 1 Codecs and Fax Jitter Buffer options added Version 1 1 February Configuration 2006 3 7 2 SIP Extensions Page SIP notify messages option added to Version 1 1 February keep the SIP proxy connection alive 2006 3 7 4 STUN Client Page The STUN Client submenu was added to Version 1 1 February the Telephone menu 2006 Voice Gateways System Manual Legal Rights Topic Description Date Issued 3 7 6 Line Configuration Page The Line Configuration submenu was added to the Telephone menu Version 1 1 February 2006 3 9 5 RTP Stats Page The RTP Statistics submenu was added to the System menu Version 1 1 February 2006 3 10 Upgrade Page Download option from an HTTP server was added Version 1 1 February 2006 3 12 Logout Page Logout option added Version 1 1 February 2006 General No H323 support Version 1 2 August 2006 2 3 Installation and Commissioning and 3 2 Accessing the Web Configuration Server 2 3 Installation and Commissioning Login with user name and password Added access to the VG via LAN in addition to WAN using the WAN IP Version 1 2 August 2006 Version R2H276 December 2006 3 5 1 WAN Status Page and 3 5 2 WAN Configuration Page Broadcast Limit and Multicast Limit deleted Version R
23. 11U T38 Figure 3 12 SIP Configuration Page VG 1D2V Voice Gateways System Manual 39 Chapter 3 Using the Web Configuration Server H323 Telephone page Line Configuration Telephone Dialplan fix Tlx Dial Timeout seconds a O Use as a quick dial function Line 1 Telephone Line on Cot HA Mode Gate Keeper IP primary Gate Keeper IP secondary H323 Alias Outgoing Display Name Telephone Number Incoming CLIP Keepalive Timeout seconds Ring Signal 0 9 Preferred Codecs Set Codecs Fax save Cancel C Fixed puto oft m i Danny 0720310004 Con of heo ooo eooo 6729 GTA G711U T38 Figure 3 13 H323 Telephone Page VG 1D2V The SIP Configuration page H323 Telephone pages include the following components 40 Operation Telephone Menu Table 3 6 SIP Configuration H323 Telephone Page Parameters Parameter Description Dialplan The Dialplan parameter defines how the Voice Gateway decides that a complete number has been dialed See more details in Section 3 7 7 The default value is xx T xx which means that each of the following schemes can be used E xx T Dial timeout Any number of digits may be dialed Following T seconds in which no new digit is dialed a decision is reached that dialing was completed and the unit will send the dial
24. 2H276 December 2006 3 7 1 SIP H323 Configuration Page Default dialplan changed Version R2H276 December 2006 3 7 1 1 Codecs and Fax Configuration Optional use of G711A U codex enabled Version R2H276 December 2006 3 7 7 1 Hotline Hotline option added to the dialplan Version R2H276 December 2006 3 7 7 2 Adding Removing Prefixes Automatic addition and removal of prefixes options added to the dialplan Version R2H276 December 2006 Voice Gateways System Manual Legal Rights Legal Rights Copyright 2006 Alvarion Ltd All rights reserved The material contained herein is proprietary privileged and confidential and owned by Alvarion or its third party licensors No disclosure thereof shall be made to third parties without the express written permission of Alvarion Ltd Alvarion Ltd reserves the right to alter the equipment specifications and descriptions in this publication without prior notice No part of this publication shall be deemed to be part of any contract or warranty unless specifically incorporated by reference into such contract or warranty Trade Names Alvarion BreezeCOM WALKair WALKnet BreezeNET BreezeACCESS BreezeMANAGE BreezeLINK BreezeCONFIG BreezeMAX AlvariSTAR BreezeLITE AlvariCRAFT MGW eMGW WAVEXpress MicroXpress WAVEXchange WAVEView GSM Network in a Box and TurboWAVE and or othe
25. 3 5 1 WAN Status Page Home WAN Status WAN Status Interface Status Enabled Yes Service Bridged Bridge Status Forwarding Upgrade Protocol Ethernet pee Interface Status Up Li gout Network Settings Dynamic IP Assignment YES via DHCP IP Address 192 168 2 1 MAC Address 00 0f 5d fe 7d 50 Subnet Mask 255 255 255 0 Default Gateway 192 168 2 254 DNS Address 172 17 10 141 Domain Name VLAN Tag Not set Priority Tag Not set _Uruiete Figure 3 3 WAN Status Page The WAN Status page includes the following components 22 Operation WAN Menu Table 3 2 WAN Status Page Parameters Parameter Description Interface Status Bridge Status Enabled The administrative status of the WAN port Yes or No In the current version the administrative status cannot be disabled Service The configured operation mode In current version it is always Bridged The method of handling packets with an unknown destination address In the current version it is always Forwarding Protocol The protocol used for data transmission In the current version it is always Ethernet Interface Status The operational status of the WAN port Up or Down Network Settings Dynamic IP Assignment The method of configuring IP Address Subnet Mask Default Gateway and DNS Address as defined in the WAN Configuration page E Yes via DHCP the parameters are obtained from a DHCP server E No the
26. 3 9 2 Localization Page deo ecw tciataceteinccteiows cats Sectactvdasyuauesteectdencusyadascedeemiauasameteaced 64 3 9 3 SNMP Configuration Page siiisctescstabnidesctascetgubecrdeeneAoeneisstnenel nioeaiaiteeaaaaaaen ade 65 3 9 4 Service Access Configuration Page ccccccccesesseeeeeeeneeeeeeesaeeeteeseeeeeeeaaeeees 66 39 5 RTP Statistics Page scccisactsccsdsasca betes gndeeecatnesstenstchoniatelvsrnnpas annsgdeea EE 67 3 10Upgrade PAG C viisiicnseseninnetsratasatieiwnsiennatnnndasdinanceantiperinneteeannna ie ane 68 3 10 1 Downloader Result Codes hexadecimal ccccceceeeeeeeeeeeeeeseneeeeeseeeeesenaeeees 69 3 11Restart PAG ti wiiinsansetinasieaancnasasnannunediaidetenansaatainairnsiua ands ieumuadacnadauadndinannaenmaiinedtaeaaste 71 3 12Logo t Page sce siaiscssenasisnncintaneanicdswncereundinaanals snnteaantaxananateninzeserasadaaabcenmnnnnnaddinerenneaicianins 72 3 1 SParameters SUMMMALY sasseciencecsidencstecntacsceteedencenticgeenceannerine ace dinaanan tanaka kesk asain 73 Appendix A Internal Class 5 Services A 1 Actions and Keypad Sequences csseccceseeeeeeeeeeeeeeesseeeeeeeeeeaeeeeeeeeneeeesseeaeaeenseeneeeees 80 A 2 Using the Class 5 Sry eS cited iaseccesstticec teens cece nses ot vest acids cece eet acecepeteisaeenaticaes 81 PZ NV Call WV ANN seosiaxicss cies ctacaasene Mavsabeuces E enn eeeap ates 81 A 2 2 Call Inquiry 22icsai anceinocetanteeconcassonas te cderaecenasinasedaatsd acesd tagicenmracaneieniaasenedenieienann
27. Auto Crossover is used to automatically identify the cable type and internally switch over the TX RX paired signals MIB Management Information Base A database of objects that can be monitored by a network management system SNMP uses standardized MIB formats that allow any SNMP tools to monitor any device defined by a MIB NAT Network Address Translation An IETF standard that allows an organization to present itself to the Internet with far fewer IP addresses than there are nodes on its internal network The NAT technology which is typically implemented in a router converts private IP addresses such as in the 192 168 0 0 range of the machine on the internal private network to one or more public IP addresses for the Internet It changes the packet headers to the new address and keeps track of each session When packets come back from the Internet NAT performs the reverse conversion to the IP address of the client machine NTP Network Time Protocol A protocol used to update the real time clock in a computer There are numerous primary and secondary servers in the Internet that are synchronized to the Coordinated Universal Time UTC via radio satellite or modem For more information visit www ntp org POTS Plain Old Telephone System A basic analog telephone equipment Voice Gateways System Manual 93 Glossary PPPoE REN RH RTCP RTP SIP 94 Point to Point Protocol over Ethernet PPPoE relies on two
28. Ethernet cable should not exceed 100m Use a straight cable for connecting to a PC or a crossed cable for connecting to a hub switch Installation Installation and Commissioning 13 Use standard telephone cord s with RJ 11 termination to connect the telephony equipment to the unit 14 Verify proper operation using the LED indicators see Table 2 2 15 To verify data connectivity from the end user s PC or from a portable PC connected to the unit try to connect to the Internet or to ping another unit in the network 16 Verify proper telephony operation by establishing a call to another telephone for each enabled line Voice Gateways System Manual 13 Chapter 2 Installation 2 4 Notes on Using the Voice Gateways in Alvarion s Systems 2 4 1 BreezeMAX System Version 1 5 and higher Access the Monitor program of the SU from a PC connected to the LAN port of the Gateway The SU s Monitor program uses the fixed IP address 192 168 254 251 with the subnet mask 255 255 255 0 The PC used for accessing the Monitor program should be configured to belong to the same subnet It is recommended to set the PC s IP address to 192 168 254 250 which is the default TFTP Server IP address in the Monitor required for downloading SW versions and for downloading uploading configuration files Information about the DRAP enabled Gateways that are connected to each SU can be viewed in the Base Station s Monitor program in the Voice
29. Explorer Jitter Buffer Adaptive Jitter Buffer 100 Max playout delay lt 300mS Fixed Jitter Buffer ao Fixed playout delay lt 120mS I Automatically switch to Fixed Jitter Buffer upon faxmodem tone detection Line 1 Codec Packet Keypad Priority 30 RFC2833 j Vv o7i1a O zo None 7 2 Vv e7110 O zo None x 3 I 138 Fax Line 2 Codec Packet Keypad Priority 30 RFC2833 Vv o711a O zo None 7 2 Vv e7110 TT zo None x s I 138 Fax Figure 3 14 Codecs and Fax Configuration Window VG 1D2V Voice Gateways System Manual 45 Chapter 3 Using the Web Configuration Server The jitter buffer options are common to both lines if applicable Table 3 7 Jitter Buffer Options Parameter Description Adaptive Jitter Buffer Maximum Delay The Voice Gateway uses a Jitter Buffer to eliminate jitter effects The size of the buffer changes dynamically to reflect actual jitter conditions The Adaptive Jitter Buffer Maximum Delay defines the maximum size that is available for the jitter buffer the larger the size the greater the potential delay The range is from 100 to 300 milliseconds The default duration is 100 milliseconds Fixed Jitter Buffer When using fax only it is recommended to use a fixed jitter buffer The fixed jitter buffer may affect voice conversation performance The range is from 100 to 300 milliseconds The default duration is
30. HIRD PARTIES ALVARION NEITHER ASSUMES NOR AUTHORIZES ANY OTHER PERSON TO ASSUME FOR IT ANY OTHER LIABILITY IN CONNECTION WITH THE SALE INSTALLATION MAINTENANCE OR USE OF ITS PRODUCTS Limitation of Liability a ALVARION SHALL NOT BE LIABLE TO THE PURCHASER OR TO ANY THIRD PARTY FOR ANY LOSS OF PROFITS LOSS OF USE INTERRUPTION OF BUSINESS OR FOR ANY INDIRECT SPECIAL INCIDENTAL PUNITIVE OR CONSEQUENTIAL DAMAGES OF ANY KIND WHETHER ARISING UNDER BREACH OF CONTRACT TORT INCLUDING NEGLIGENCE STRICT LIABILITY OR OTHERWISE AND WHETHER BASED ON THIS AGREEMENT OR OTHERWISE EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGES b TO THE EXTENT PERMITTED BY APPLICABLE LAW IN NO EVENT SHALL THE LIABILITY FOR DAMAGES HEREUNDER OF ALVARION OR ITS EMPLOYEES OR AGENTS EXCEED THE PURCHASE PRICE PAID FOR THE PRODUCT BY PURCHASER NOR SHALL THE AGGREGATE LIABILITY FOR DAMAGES TO ALL PARTIES REGARDING ANY PRODUCT EXCEED THE PURCHASE PRICE PAID FOR THAT PRODUCT BY THAT PARTY EXCEPT IN THE CASE OF A BREACH OF A PARTY S CONFIDENTIALITY OBLIGATIONS Disposal of Electronic and Electrical Waste Disposal of Electronic and Electrical Waste Pursuant to the WEEE EU Directive electronic and electrical waste must not be disposed of with unsorted waste Please contact your local recycling authority for disposal of this product vi Voice Gateways System Manual Legal Rights Important Notice This user manual is delivered subject to the follow
31. Otherwise check the No option Note that only one VLAN can be untagged on each port or on both 4 Click OK The newly added entry will be added to the Tagged Port Membership table gt To delete a VLAN from the Tagged Port Membership table 1 Click on the row ID number of the entry you wish to remove The VLAN Editor Delete is displayed 28 Operation VLAN Tagging Menu VLAN Editor Microsoft Internet Explorer Row number 1 VLAN ID 1 4094 ooos VLAN NAME fvoicecung VLAN priority o 7 ves C No ves C No Figure 3 7 VLAN Editor Delete VLAN 2 Click on the Delete button The entry will be removed from the Tagged Port Membership table 3 6 3 VoIP VLAN Configuration Page Telephone By Reservation system Upgrade Restart Logout VoIP VLAN Configuration VoIP VLAN Configuration Call Signaling VLAN Tag Priority Tag RTP VLAN Tag Priority Tag Save VolP VLAN Settings Figure 3 8 VoIP VLAN Configuration Page Voice Gateways System Manual Chapter 3 Using the Web Configuration Server The VoIP VLAN configuration page enables defining the following parameters Table 3 5 VoIP VLAN Configuration Page Parameters Parameter Description Call Signaling VLAN Tag The VLAN ID tag for VoIP call signaling packets If not set the Default VLAN ID set for WAN in the VLAN Tagging page will also apply for VOIP Priority Tag T
32. PPLICATIONS WHERE THE FAILURE MALFUNCTION OR INACCURACY OF PRODUCTS CARRIES A RISK OF DEATH OR BODILY INJURY OR SEVERE PHYSICAL OR ENVIRONMENTAL DAMAGE HIGH RISK ACTIVITIES HIGH RISK ACTIVITIES MAY INCLUDE BUT ARE NOT LIMITED TO USE AS PART OF ON LINE CONTROL SYSTEMS IN HAZARDOUS ENVIRONMENTS REQUIRING FAIL SAFE PERFORMANCE SUCH AS IN THE OPERATION OF NUCLEAR FACILITIES AIRCRAFT NAVIGATION OR COMMUNICATION SYSTEMS AIR TRAFFIC CONTROL LIFE SUPPORT MACHINES WEAPONS SYSTEMS OR OTHER APPLICATIONS REPRESENTING A SIMILAR DEGREE OF POTENTIAL HAZARD ALVARION SPECIFICALLY DISCLAIMS ANY EXPRESS OR IMPLIED WARRANTY OF FITNESS FOR HIGH RISK ACTIVITIES b PURCHASER S SOLE REMEDY FOR BREACH OF THE EXPRESS WARRANTIES ABOVE SHALL BE REPLACEMENT OR REFUND OF THE PURCHASE PRICE AS SPECIFIED ABOVE AT ALVARION S OPTION TO THE Voice Gateways System Manual v Legal Rights FULLEST EXTENT ALLOWED BY LAW THE WARRANTIES AND REMEDIES SET FORTH IN THIS AGREEMENT ARE EXCLUSIVE AND IN LIEU OF ALL OTHER WARRANTIES OR CONDITIONS EXPRESS OR IMPLIED EITHER IN FACT OR BY OPERATION OF LAW STATUTORY OR OTHERWISE INCLUDING BUT NOT LIMITED TO WARRANTIES TERMS OR CONDITIONS OF MERCHANTABILITY FITNESS FOR A PARTICULAR PURPOSE SATISFACTORY QUALITY CORRESPONDENCE WITH DESCRIPTION NON INFRINGEMENT AND ACCURACY OF INFORMATION GENERATED ALL OF WHICH ARE EXPRESSLY DISCLAIMED ALVARION WARRANTIES HEREIN RUN ONLY TO PURCHASER AND ARE NOT EXTENDED TO ANY T
33. Session Initiation Protocol SIP sessions by sending repeated INVITE requests The repeated INVITE requests or re INVITEs are sent during an active call leg to allow user agents UA or proxies to determine the status of a SIP session Without this keep alive mechanism proxies that remember incoming and outgoing requests stateful proxies may continue to retain call state needlessly If a UA fails to send a BYE message at the end of a session or if the BYE message is lost because of network problems a stateful proxy does not know that the session has ended The re INVITES ensure that active sessions stay active and completed sessions are terminated The range for the timer value is from 1 to 999 seconds Use NOTIFY message to keep alive the session with SIP proxy every X seconds The gateway will send a SIP NOTIFY message to the SIP proxy at the configured interval These messages can keep the connection with SIP proxy alive as well as the NAT port mapping when the Voice Gateway is behind NAT The range is 0 99999 Default interval 15 seconds Voice Gateways System Manual Chapter 3 Using the Web Configuration Server Ce Click on the Save SIP Extensions Settings button before leaving the page to save the new settings The new settings will be applied after restarting the unit NAT Traversal Configuration Page SIP Only NAT Traversal function can be used to allow the Voice Gateway to registe
34. TF IP HyperText Transport Protocol A protocol used to request and transmit files especially web pages and web page components over the Internet or other computer network Indoor Unit International Electrotechnical Commission Geneva Switzerland www iec ch An organization that sets international electrical and electronics standards founded in 1906 It is made up of national committees from over 60 countries Institute of Electrical and Electronics Engineers IEEE pronounced I triple E is an organization composed of engineers scientists and students The IEEE is best known for developing standards for the computer and electronics industry In particular the IEEE 802 standards for local area networks are widely followed A QoS method A three bit value that can be placed inside an 802 1Q frame tag The IEEE 802 1Q standard defines the operation of VLAN Bridges that permit the definition operation and administration of Virtual LAN topologies within a Bridged LAN infrastructure The 802 1Q specification establishes a standard method for inserting VLAN membership information into Ethernet frames A tag field containing VLAN and or 802 1p priority information can be inserted into an Ethernet frame carrying VLAN membership information Internet Engineering Task Force One of the task forces of the IAB Internet Architecture Board formally called the Internet Activities Board which is the technical body that oversees th
35. TUN Client 93 ToS DiffServ Line Configuration STUN Client Configuration Note Static NAT traversal shall be turned off if STUN client is enabled in this page STUN Client Mode C on off STUN Server AddresxIP or Domain STUN Serer Port 2478 Nat Type External IP Address Save STUN Client Settings Figure 3 17 STUN Client Configuration Page The STUN Client page includes the following components Table 3 11 STUN Client Configuration Page Parameters STUN Client Mode Switch the STUN Client mode on or off When on turn off the Static NAT Traversal mode STUN Server Address IP or Domain The IP address or Domain of the STUN server STUN Server Port The port used by the STUN Server The default is port 3478 Click on the Save STUN Client Settings button before leaving the page to save the new settings The new settings will be applied after restarting the unit STUN enabled cannot operate with NAT traversal enabled In any case the Voice Gateway receives the external IP and the port information using the STUN 3 7 5 ToS Page Outgoing packets from the Voice Gateway can be marked with DSCP DiffServ Code Point values The ToS page enables defining the 8 bits ToS field in the IP header for different packet types Diffserv use the first 6 out of these 8 bits For more information about DiffServ Code Points please refer to RFC2474 52 Operation Telephone Menu
36. a a reed We re on your wavelength Voice Gateways SW Version SIP R2H276 H323 R2F194 December 2006 P N 214272 Document History Document History Topic Description Date Issu ed 1 2 8 Connectors The connectors specifications were updated Version 1 0 November 2005 2 3 Installation and Commissioning description added Version 1 0 November Commissioning 2 4 Notes on Using the Voice Using the VG with BreezeMAX and Version 1 0 November Gateway VG in Alvarion s BreezeACCESS VL Systems 3 2 Accessing the Web Detailed instructions for accessing the Version 1 0 November Configuration Server web server via the WAN and via the LAN were added 3 5 1 WAN Status Page Bridge Status new display Version 1 0 November 3 6 5 VLAN Configuration The example was updated Version 1 0 November Example 2 3 7 Telephone Menu The Telephone Menu was modified to Version 1 0 November include the H323 Configuration page 3 12 Web Configuration Some of the parameters defaults have Version 1 0 November Server s Parameters Summary been changed also in their respective paragraphs 2 3 Installation and Updated for VG management via WAN Version 1 1 February Commissioning only 2006 3 6 1 VLAN Tagging Page Default VLAN ID for LAN was removed Version 1 1 February and explanations were added 2006 3 6 6 VLAN Configuration An additional example was provided Version 1
37. and show the number without the prefix on the phone display Use the following notation lt replacement string received string gt For example Set the dialplan to lt 8 1860 gt xxx When a number with the 1860 prefix is received e g 1860123 the prefix 1860 is replaced with the digit 8 and the number displayed is 8123 Voice Gateways System Manual 57 Chapter 3 Using the Web Configuration Server 3 8 BW Reservation DRAP Configuration Page The Voice Gateway uses DRAP Dynamic Resource Allocation Protocol for efficient management of bandwidth resources for telephone calls DRAP Configuration Home DRAP Server Settings 7 Enable DRAP O Enable Pre allocation DRAP Server IP Address Server Port e171 DRAF Protocol Options Discovery Time S fic Acknowledge Time 100mS E Clear Count ze Retry Count s RTP Packing Ratio sons Vocoder Type e729 Save DRAP Settings ogout Figure 3 20 DRAP Configuration Page The DRAP Configuration page includes the following components 58 Operation BW Reservation DRAP Configuration Page Table 3 13 DRAP Configuration Page Parameters Parameter Description DRAP Server Settings Enable DRAP The Enable DRAP option defines whether DRAP is used for establishing telephone voice and fax calls If enabled a DRAP Server must be available to provision telephone calls The default is disabled unch
38. another page menu item or before quitting the application The Save Settings functionality in many cases is per page if you leave the page without clicking the Save Settings button all the changes in the page will be lost Changes to most of the settings are applied only after restarting the unit refer to Section 3 11 18 Operation Using the Web Configuration Server CAUTION A There is no control that the entered values are valid or have the correct format or range If invalid values are entered access to the unit may be lost and in that case a factory default procedure must be performed Refer to Section 2 2 2 for information about how to reset the Voice Gateway to factory default parameters Voice Gateways System Manual 19 Chapter 3 Using the Web Configuration Server 3 4 Home Menu Product info Page The Product Info page provides general information on the Voice Gateway Logout Product info Name Mac address Serial number Product number Product revision Production week Default configuration Downloader revision Reported download status Main software revision Operator defaults revision Figure 3 2 Product Info Page VG 1D1V 00 0f 5d fe 7d bc NA NA RIA 05w25 RIA exe_132_4888 R1B12 0x0 DMA0027 R2F201 local_ringing_off ini The Product Info page includes the following components Table 3 1 Product Info Page Parameters Name The unit s model Mac add
39. are designed for remote management and supervision using either the built in internal web server or SNMP The Voice Gateways are easily updated and upgraded as they support remote software and configuration file download System Description Specifications 1 2 1 2 1 Table 1 1 Specifications Telephony and Fax Services Telephony and Fax Services Item VoIP Standard Description HM H323 model H323v2 4 E SIP model SIP RFC 3261 Internal Class 5 Services Call Waiting 3 party call call hold and call alteration differentiated ringing tones refer to Appendix A for more details External Class 5 Services Activation deactivation of class 5 services supported by the IP telephony system Fax G3 compliant V 17 14 4 Kbps fax reception and transmission using the T 38 standard or in band using G 711 codec CNI rerouting Calling Number Identification 3rd party initiated pause and FSK DTMF External rerouting of media stream during speech e g for pre paid calling card and record announcement DTMF In band and out band using H 245 and H 225 Regional Settings Telephony signals tones and cadences see Appendix B 1 2 2 Security Table 1 2 Security Item Description VLAN Support IEEE 802 1Q with up to 16 VLAN IDs Authentication Per call authentication and registration Voice Gateways System Manual
40. ata transmission TCP is the part of the TCP IP suite of protocols that is responsible for forming data connections between nodes that are reliable as opposed to IP which is connectionless and unreliable Transmission Control Protocol Internet Protocol A set of protocols developed by the U S Department of Defense to allow communication between dissimilar networks and systems over long distances TCP IP is the de facto standard for data transmission over networks including the Internet Trivial File Transfer Protocol Simplified version of FTP that allows files to be transferred from one computer to another over a network usually without the use of client authentication Type Of Service A field in an IP packet IP datagram that is used for quality of service QoS User Datagram Protocol Connectionless transport layer protocol in the TCP IP protocol stack UDP is a simple protocol that exchanges datagrams without acknowledgments or guaranteed delivery requiring that error processing and retransmission be handled by other protocols UDP is defined in RFC 768 Voice Gateways System Manual 95 Glossary URL V 29 V 32bis VLAN VoIP 96 Uniform Resource Locator the global address of documents and other resources on the World Wide Web The first part of the address indicates what protocol to use and the second part specifies the IP address or the domain name where the resource is located An ITU fax
41. ation Parameters T38 Fax The T38 check box indicates for each line whether to support the T38 Fax protocol The default is checked T38 Fax supported Click on the Save button before leaving the page to save the new settings The new settings will be applied after restarting the unit 3 7 2 SIP Extensions Page Home SIP Extensions NAT STUN Client ToS DiffSery Line Configuration SIP Extensions I Support PRACK method with provisional response reliability I Encode SIP URI with user parameter I Encode default port in SIP URI M Include default port in INVITE I Send INVITE with Timer header value D I SIP Session Timer value Logout MV Use NOTIFY message to keep alive the session with SIP proxy every 15 seconds Save SIP Extension Settings Figure 3 15 SIP Extensions Page The SIP Extensions page includes the following components Table 3 9 SIP Extensions Page Parameters Support PRACK method The PRACK request plays the same role as ACK but for provisional with provisionalresponse responses PRACK is a normal SIP message like BYE As such its own reliability reliability is ensured hop by hop through each stateful proxy Also like BYE but unlike ACK PRACK has its own response If this were not the case the PRACK message could not traverse proxy servers compliant to RFC 2543 For more details refer to RFC 3262 Reliability of Provisional Responses in the Session Initiation Protoco
42. by and x T means none or at least one digit followed by T However having only T in the dialplan where x is null activates the Hotline function see Section 3 7 7 1 To avoid unwanted activation of the hotline function use the default dialplan XX XX T Simple dialplan Example 1 Following example allows dialing any 7 digit number e g 5551234 or an operator on 0 Dialplan is OT xxxxxxx Complex dialplan Example 2 Local operator on O long distance operator on 00 four digit local extension number starting with 3 4 or 5 seven digit local numbers are prefixed by an 8 two digit star services e g 69 ten digit long distance prefixed by 91 and international numbers starting with 9011 one or more digits Dialplan for this is OT OOT 3 5 xxx 8xxxxxxx xx 9 1xxxxxxxxxx 9011xx T Voice Gateways System Manual 55 Chapter 3 Using the Web Configuration Server 3 7 7 1 3 7 7 2 56 Using supplemental external services Example 3 When a soft switch or a SIP server gatekeeper exists in the network and the user would like to use class 5 services which are not internal to the Voice Gateway e g xy xy abcd xy etc the VG dialplan should be configured as follows 0 9 0 9 Note that when VG internal class 5 services are enabled default in addition to the above dialplan the internal class 5 activation codes remain valid See Appendix A Call completion Call co
43. ck on the Refresh button The version R2H implementation of the dialplan module is fully compliant with RFC3435 However some previously acceptable dialplans are inconsistent with RFC3435 operation and must therefore be changed to ensure continued operation consistent with previous releases You should review your dialplan to ensure that the dial behavior will continue to be that required by your customers For example the default dialplan should be changed from x x T to xx xx T It is recommended to review your dialplan to ensure that its proper operation You can change the dialplan as follows E Default dialplan Ifan INI file is not used upgrade to version R2H and then install the INI file provided with the upgrade package Ifan INI file is used you will be provided with an updated INI file E Non default dialplan Ifan INI file is not used Refer to Section 3 7 7 for information on updating your dialplan The dialplan can be changed via the Telephone menu or via option 43 If an INI file is used you will be provided with an updated INI file 3 10 1 Downloader Result Codes hexadecimal If something goes wrong during download or installation you will be notified according to the following HM O 0x00 normal boot no upgrade requested or needed E bit O 0x01 upgrade requested or main application not valid E bit 1 0x02 failed to download new image E bit 2 0x04 TFTP server not defined E b
44. d LAN LEDs bilnk twice indicating that the unit operates properly Then the PWR LED is lit Other LEDs may also be lit according to the status of the WAN LAN and Phone ports as described in Section 2 2 3 4 Connect a PC to the WAN or LAN port using a crossed Ethernet cable Configure the PC with a static IP address 192 168 254 2 and subnet mask 255 255 255 0 The IP address of the WAN port for management purposes only is 192 168 254 254 and netmask 255 255 255 0 NOTE x The VG can be accessed via the WAN or LAN port using the WAN IP address 5 Open a web browser and connect to the unit by entering http 192 168 254 254 in the address field Voice Gateways System Manual 11 Chapter 2 Installation 6 10 11 12 If the Web Configuration Server is password protected you will be prompted to enter your username and password in order to log in to the system The default username is operator and the default password is installer See Chapter 3 for details on using the Web Configuration Server Configure the necessary parameters according to instructions supplied by the system administrator The mandatory parameters that must be configured properly are Enable DRAP in BW Reservation page only if DRAP is supported by the wireless system currently DRAP is supported by BreezeMAX equipment with SW version 1 5 or higher and BreezeACCESS VL with SW version 4 0 Uncheck if DRAP is not used LAN WAN VLAN Tagged Por
45. d scalability When a secondary server is available the unit will try to register to the secondary server after 10 failed attempts to register to the primary server The available options are E Fixed The secondary Gate Keeper SIP Server IP address is defined manually by the Gate Keeper SIP Server IP secondary parameter E Auto The secondary Gate Keeper SIP Server IP address is supplied by the primary Gate Keeper SIP Server E Off Secondary Gate Keeper SIP Server is not supported The default is Off SIP Server IP primary SIP model only SIP Server Port primary SIP model only The IP address for the primary SIP server proxy who is responsible for managing the Voice Gateway in the specific network If HA mode is set to Auto the primary SIP server proxy provides to the Voice Gateway during registration an IP address for the secondary system The port used for the primary system The recommended values are in the range 1030 65536 The default is 5060 SIP Server IP secondary SIP model only The IP address of the secondary SIP server proxy SIP Server Port secondary SIP model only The port used for the secondary system The recommended values are in the range 1030 65536 The default is 5060 Gate Keeper IP primary H323 model only The IP address for the primary Gate Keeper who is responsible for managing the Voice Gateway in the specific network If HA mode is set to Auto
46. ditions is met a IP address Private IP address b received parameter in INVITE or REGISTER IP address is not equal to internal IP address E Off NAT Traversal function is disabled The default is Off When using a NAT device it is recommended to set this parameter to ON and to enter the External NAT mapped IP Address 3 7 4 Click on the Save button before leaving the page to save the new settings The new settings will be applied after restarting the unit STUN Client Configuration Page SIP only Simple Traversal of UDP STUN is a method of NAT traversal through UDP based on RFC 3489 For proper voice conversations in networks based on more than one NAT the NAT devices must be able to transfer RTP packets in both directions The STUN protocol uses the default server port 3478 The Voice Gateway communicates with a server over the internet Based on the RTP packets the Server knows the number of NATs behind which the Voice Gateway is located the IP address of the Voice Gateway and which of the Voice Gateway s ports are actually used The STUN application supports the following NAT Types FullCone FullRestrict and PortRestrict The Voice Gateway does not support symmetrical NAT as it is not meant to traverse symmetrical NAT devices Voice Gateways System Manual 51 Chapter 3 Using the Web Configuration Server Home VLAN Telephone ervation Restart Logout SIP SIP Extensions NAT S
47. dium for telephone calls by sending voice data in packets using IP rather than by traditional circuit switched transmissions of the PSTN Voice Gateways System Manual Glossary WAN Wide Area Network A computer network that spans a relatively large geographical area Wide area networks can be made up of interconnected smaller networks spread throughout a building a state or the entire globe Voice Gateways System Manual 97
48. do forward to lt telephone number gt Indication tone is heard Deactivate Call Forward Voice Gateways System Manual 83 Appendix B Default Telephony Parameters In This Appendix This appendix provides the default settings for various telephony parameters Appendix B Default Telephony Parameters Table B 1 Default Telephony Parameters Parameter Definition Default Normal Ringing Signal The signal that end user will hear from the telephone set when a Call is received Cadence 1 second on 4 second off Duration 180 seconds Frequency 25 Hz originates a call hears when the destination telephone line is busy Ringing Tone The tone that sounds on the Cadence 1 second on 5 telephone set when ringing on second off the other side ae Duration Not limited Frequency 425 Hz Level 10 dBmO Dial Tone The tone that the call originator Cadence Continuous hears in the handset before Durata Katit ration imite dialing the destination telephone pie number Frequency 425 Hz Level 5 dBmO Busy Tone The tone that the end user that Cadence 0 25 second on 0 25 second off Duration Not limited Frequency 425 Hz Level 10 dBmO Network Busy Tone The tone that the end user that originates a call will hear when the network is congested Cadence 0 25 second on 0 75 second off Duration Not limited Frequency 425 Hz Level 10 dBmO
49. e development of the Internet suite of protocols commonly referred to as TCP IP The IETF is responsible for solving short term engineering needs of the Internet Internet Protocol The standard that defines how data is transmitted over the Internet IP bundles data including e mail faxes voice calls and messages and other types into packets in order to transmit it over public and private networks Voice Gateways System Manual 91 Glossary ISO ITU T IVR LAN MAC MAC Address 92 International Organization for Standardization Geneva www iso ch An organization that sets international standards founded in 1946 The U S member body is ANSI ISO deals with all fields except electrical and electronics which is governed by the older International Electrotechnical Commission IEC With regard to information processing ISO and IEC created JTC1 the Joint Technical Committee for information technology International Telecommunication Union Telecommunications An intergovernmental organization through which public and private organizations develop telecommunications The ITU was founded in 1865 and became a United Nations agency in 1947 It is responsible for adopting international treaties regulations and standards governing telecommunications The standardization functions were formerly performed by a group within the ITU called CCITT but after a 1992 reorganization the CCITT no longer exists as
50. ecked Enable Pre allocation The Enable Pre allocation option defines whether resource allocation is requested immediately upon off hook condition or only after dialing the requested number When disabled unchecked a request for resource allocation will be sent only after dialing the number When enabled the resource allocation request will be sent immediately and a dial tone will be provided only if the requested resources are available The default is enabled checked DRAP Server IP Address The IP address of the DRAP server that should serve the resource allocation requests of the unit Leave empty for Auto Discovery The default is an empty field Auto Discovery Server Port The UDP port used for the DRAP server The port number indicated will be used for originating ALLOC messages and the port number indicated 1 will be used for receiving CONFRM messages The available range is from 8000 to 8200 The default is 8171 DRAP Protocol Options Discovery Time The Discovery Time is the timeout to be used when the Auto Discovery process is used for finding a DRAP server The Auto Discovery process is based on sending empty broadcast allocation requests and the Discovery Time is the time that the unit will wait for a response before sending a new request The range is 1 to 255 seconds The default is 10 seconds Voice Gateways System Manual 59 Chapter 3 Using the Web Configura
51. eeeeees 14 Contents 2 4 1 BreezeMAX System Version 1 5 and higher cccccceeeeeeeeeeeeeeeeeeeeeeeeseeeeees 14 2 4 2 BreezeACCESS VL System Version 3 1 cceececceeeeeseeceeeeseeeeeeeeeeeeeeeeaeeeeee 14 Chapter 3 Using the Web Configuration Server 3 1 Introduction to the Web Configuration Server ecccessseceeeeeeeeeeeseeeeeeeseeeeeeeeees 16 3 2 Accessing the Web Configuration Servel cs seccsseseeneeeeseeeeeeeeeeeesesneeeeeeeeneneneeeees 17 3 3 Using the Web Configuration Server cccsecccssseenesseneneeeeeseneeeenseeeeeesseeneeeessenneeeees 18 3 4 Home Menu Product info Page s eeeeceeeeesseeeeeeeeeeeeneeeeeeeeneeeeeeeseeeseeeeeeeeeeeeennens 20 3 5 WAN Menu sesiisies insti nicennnetecararetaeanawnsiernnsetananeenaneentienniiernninenaaennndalen nanan k anaana 22 3 5 1 WAN Status Page sisi dcnvaotixcdiscciavexionaiioanaaidenwelvenannandelaawadbeniann bo EE EE 22 3 5 2 WAN Configuration Page 2cicscczsccsertavseecartucasaeniacedisendeinneececsatidacd eammauenmianeen 24 3 6 VLAN Tagging Menu s sssssunssennnnnnunnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnn nnnm nnmnnn nnmnnn 26 3 6 1 VLAN Tagging Page a aden tecnaecccutccihccevtaxsgnsstennetdeuaendnsecednelsanancanectdenqueestarseuantdner 26 3 6 2 Adding and Deleting VLANS s ssssssssnssssssssnsneenesosesnrnnnnnnerrnnnnnnnnnnnnnnnnnnneneeeeenan 27 3 6 3 VoIP VLAN Configuration Page scccemscvececedcaasdsccrescessenceineeutea
52. ephony system Phone 2 Phone service E Off Phone line does not get IP telephony VG 1D2V only indication services E On Phone line is connected to the IP telephony system LAN LAN port status E Off Ethernet Link not detected fore indication E On Ethernet link connected no activity E Blinking Ethernet link activity WAN WAN port status E Off Ethernet link not detected indication E On Ethernet link connected no activity E Blinking Ethernet link activity POWER PWR Power Indication E Off unit is not powered or power failed E Green power OK Installation Installation and Commissioning Zu Installation and Commissioning The unit can be placed on a desktop or a shelf The location should be selected taking into account the necessary connections to mains power SU IDU and user s data telephony equipment It is assumed that installation and commissioning of the SU has already been completed and that the SU is connected to the Base Station gt To install the Voice Gateway 1 Connect the DC power cable of power supply to the 12 VDC jack on the rear panel of the unit 2 Connect the mains power cable to the power supply Connect the other end of the mains power cable to the AC mains NOTE x The color codes of the power cable are as follows Brown Phase Blue Neutral 0 Yellow Green Ground 3 After power up all front panel LEDs bilnk once and then the PWR WAN an
53. er s telephones LAN 10 100Base T RJ 45 Connection to the user s data equipment WAN 10 100Base T RJ 45 Connection to the SU IDU 12 VDC DC power jack Connection to power supply 2 2 2 Reset to Factory Default Configuration Press down the RESET button on the back of the unit for at least 5 seconds to reset all configurable parameters back to their original default values After releasing the RESET button the PWR WAN and LAN LEDs blink twice indicating proper operation The affect on the selected IP parameters acquisition method depends on the time the RESET button is held in the pressed position E Ifthe RESET button is pressed down for 5 to 10 seconds The unit will use DHCP to get the WAN IP parameters E Ifthe RESET button is pressed down for more than 10 seconds The unit will use the static manually defined WAN IP parameters IP 192 168 254 254 Mask 2 355 255 255 0 Voice Gateways System Manual Chapter 2 Installation For more details on configuration of DHCP and static IP parameters refer to Section 3 5 2 2 2 3 LEDs were on your wavetengh Voice Gateway x NOTE The VG 1D1V has a single Phone LED Figure 2 2 VG 1D2V Front Panel Table 2 2 Voice Gateway LEDs Name Symbol Description Functionality Phone 1 Phone service E Off Phone line does not get IP telephony a indication services E On Phone line is connected to the IP tel
54. etwork in order for the system to accept it Telephone domain name SIP model only The domain name The Telephone domain name is limited to 25 characters i e 25 characters after the sign If not specified by the user the same information as defined in the SIP Server IP field will be used Port SIP model only The number of the outgoing signaling port on the telephone line Line1 and Line 2 cannot have the same port number The range is from 1030 to 65535 The default is 5060 for Line 1 and 5061 for Line 2 Message Waiting Account SIP model only When a message is waiting in the network based voice mail system a discontinuous dial tone will be played when the handset goes off hook To enable a SIP server supporting Interactive Voice Response IVR is required Voice Gateways System Manual Chapter 3 Using the Web Configuration Server Table 3 6 SIP Configuration H323 Telephone Page Parameters Parameter Description Incoming CLIP The Calling Line Identity Presentation Caller ID option for the telephone line If On is selected the Caller ID information of a calling party in incoming calls will be displayed on a caller ID display attached to the telephone line Caller ID can be restricted permanently using a customized ini file The default is Off Keepalive Timeout seconds Ring signal 0 9 The interval of waiting for acknowledgement message
55. for untagged data 200 in the fields Untagged VLAN ID on LAN port and click Save 10 Restart the unit to apply the changes 3 6 5 VLAN Configuration Example 2 This example describes how to define the following configuration E Two daisy chained Voice Gateways VG 1 and VG 2 E VLAN ID 100 VLAN Priority 7 for Voice call signaling RTP and RTCP and Management packets on the WAN port E VLAN ID 200 VLAN Priority 4 for data on WAN port VG 1 E No VLAN for data on the LAN port VG 2 32 Operation VLAN Tagging Menu VLAN 200 VLAN100 VG 1 Figure 3 10 VLAN Configuration Example 2 3 6 5 1 VG 1 Configuration 1 Inthe VLAN page click Add VLAN to open the VLAN Editor 2 Inthe VLAN Editor enter the follwing for Voice and Management VLAN VLAN ID 100 VLAN NAME Voice amp Mng VLAN Priority 7 WAN Yes LAN Yes 3 Click OK to add the VLAN to the Tagged Port Membership table 4 Enter the VLAN ID for Voice and Management 100 in the fields Default VLAN ID on WAN port and click Save 5 In the Page Selection bar click on VoIP VLAN Configuration to open the VoIP VLAN Configuration page Enter 100 in the VLAN Tag fields for both Call Voice Gateways System Manual 33 Chapter 3 Using the Web Configuration Server 3 6 5 2 34 6 7 Signaling and RTP Enter 7 in the Priority Tag field for both Call Signaling and RTP Click Save VoIP VLAN Settings Go back to the VLAN Tagging pa
56. from the server lf Keep alive timeout is sent from the network it will override the setting in the Voice Gateway The interval for sending Keep alive registration messages from the Gateway is half the configured value 600 seconds with the default timeout of 1200 seconds In case of registration problem try changing the value to 1800 seconds The range is from 10 to 65535 seconds The default is 1200 seconds The Ring signal parameter provides a selection of 10 different ring patterns 0 9 that the unit can use The default is 0 Transport SIP model only Configure whether signaling shall use UDP or TCP The default is UDP Preferred codecs Displays the currently supported codecs according to the defined priorities Click the Set Codecs Fax button to change codecs settings priorities NOTE Click Save before clicking the Set Codecs Fax button Otherwise all configuration changes in the Telephone page will be lost Click on the Save button before leaving the page to save the new settings The new settings will be applied after restarting the unit Click the Set Codecs Fax button to change codecs settings priorities as described in the following section 44 Operation Telephone Menu 3 7 1 1 Codecs and Fax Configuration After clicking the Set Codecs Fax button the Codecs and Fax Configuration page is displayed Codecs and Fax Configuration Microsoft Internet
57. g new incoming call in progress Table A 2 Call Waiting Service Action Event RO Reject incoming call Calling party hears busy tone Continue with active call 0 R1 Disconnect on going call and answer incoming call R2 Place on going call on hold answer incoming call A 2 2 Call Inquiry Description One on going call active place a new call to a third party Table A 3 Call Inquiry Service Action Event R2 telephone number Place on going call on hold dial tone Inquire new call to a third party R1 Return to call placed on hold if third party is not answering A 2 3 Call Alteration Description Two on going calls active switch between calls Table A 4 Call Alteration Service Action Event R2 Switch between two on going calls Places non active call on hold A 2 4 Call Drop Description Two on going calls active disconnect one of the calls Voice Gateways System Manual 81 Appendix A Internal Class 5 Services Table A 5 Call Drop Service Action Event RO Disconnect call that is put on hold Continue with on going call R1 Disconnect on going call and return to call that is put on hold A 2 5 3 Party Conference 1 Description One on going call active place a new call to a third party and start conference Table A 6 3 Party Conference Service 1 Action E
58. g the table refer to Section 3 6 2 below Untagged VLAN ID The VLAN ID that is defined for untagged data on the WAN port text box on the left side and the LAN port text box on the right side This parameter must be consistent with a properly configured VLAN in the tagged port membership For examples on VLAN configuration see Section 3 6 4 and Section 3 6 5 The range for both parameters is from 1 to 4094 Default VLAN ID The text box on the left side is for the WAN port This is the VLAN defined for management frames SNMP HTTP TFTP arriving on the WAN port The DRAP packets are tagged with the default VLAN configuration The range is from 1 to 4094 NOTE x Management of the unit can only be done from the WAN port 3 6 2 Adding and Deleting VLANs gt To add a VLAN 1 Click on the Add VLAN button The VLAN Editor Add is displayed Voice Gateways System Manual 27 Chapter 3 Using the Web Configuration Server LAN Editor Microsoft Internet Explore Row number VLAN ID 1 4094 WLAN NAME WLAN priority o ves C No ves C No OK Cance1 Figure 3 6 VLAN Editor Add VLAN 2 Enter the VLAN ID 1 to 4094 VLAN NAME A descriptive string of printable characters Do not use special characters such as space or comma and the VLAN priority tag 0 to 7 3 If applicable packets need to be tagged on the WAN LAN port check the relevant Yes option
59. ge In the VLAN page click Add VLAN to open the VLAN Editor In the VLAN Editor enter the follwing for Data VLAN VLAN ID 200 VLAN NAME Data VLAN Priority 4 WAN Yes LAN Yes Click OK to add the VLAN to the Tagged Port Membership table Enter the VLAN ID for untagged data 200 in the field Untagged VLAN ID on LAN port and click Save 10 Restart the unit to apply the changes VG 2 Configuration 1 2 In the VLAN page click Add VLAN to open the VLAN Editor In the VLAN Editor enter the follwing for Voice and Management VLAN VLAN ID 100 VLAN NAME Voice amp Mng VLAN Priority 7 WAN Yes gt Oo LAN No Click OK to add the VLAN to the Tagged Port Membership table Enter the VLAN ID for Voice and Management 100 in the field Default VLAN ID on WAN port and click Save In the Page Selection bar click on VoIP VLAN Configuration to open the VoIP VLAN Configuration page Enter 100 in the VLAN Tag fields for both Call Signaling and RTP Enter 7 in the Priority Tag field for both Call Signaling and RTP Click Save VoIP VLAN Settings Go back to the VLAN Tagging page In the VLAN page click Add VLAN to open the VLAN Editor In the VLAN Editor enter the follwing for untagged data Operation VLAN Tagging Menu VLAN ID 300 an arbitrary selection a VLAN ID is required for defining the untagged data This VLAN tag is only used internally in the unit VLAN NAME Un
60. he Priority tag for VoIP call signaling packets If not set the priority tag defined for the Management VLAN in the Tagged Port Membership in the VLAN Tagging page will also apply for VOIP RTP VLAN Tag The VLAN ID tag for RTP and RTCP packets If not set the Default VLAN ID set for WAN in the VLAN Tagging page will also apply for VOIP Priority Tag The Priority tag for RTP and RTCP packets If not set the priority tag defined for the Management VLAN in the Tagged Port Membership in the VLAN Tagging page will also apply for VOIP 3 6 4 30 Typically the same VLAN is used for management call signaling and RTP In this case the same VLAN and Priority Tags should be configured for management Default VLAN on WAN port in the VLAN Tagging page Call Signaling and RTP However the Voice Gateway supports separation of VLANs and allows defining 3 different VLANs for management call signaling and RTP traffic this may require a proper router Different Priority tags for management call signaling and RTP can be configured The Priority tag for management is defined in the Priority field of the management VLAN ID configured in the Tagged Port Membership table VLAN Configuration Example 1 This example describes how to define the following configuration VLAN ID 100 VLAN Priority 7 for Voice call signaling RTP and RTCP and 5 for Management packets on the WAN port VLAN ID 200 VLAN
61. iguration Server Table 3 18 Parameters Summary Parameter Range Options Default Dialplan A string of up to 100 XX T Xx characters For details on format see Section 3 7 7 Dial Timeout 1 60 seconds 4 seconds Use Yes checked No unchecked Checked RTP Port Range Start Stop 1030 65535 Start 8000 SIP model only End 8015 Telephone Line On Off Off HA Mode Fixed Auto Off Off SIP Server IP primary IP address Null SIP model only SIP Server Port primary 1030 65535 5060 SIP model only SIP Server IP secondary IP address Null SIP model only SIP Server Port secondary 1030 65535 5060 SIP model only Gate Keeper IP primary IP address Null H323 model only Gate Keeper IP secondary IP address Null H323 model only User Name A string of up to 25 characters Null SIP model only Password A string of up to 25 characters Null SIP model only Outgoing Display Name A string of up to 25 characters Null Telephone number A string of up to 25 characters Null 74 Operation Parameters Summary Table 3 18 Parameters Summary Parameter Range Options Default H323 Alias A string of up to 25 characters Null H323 model only Telephone domain name A string of up to 25 characters Null SIP model only Port 1030 65535 Line 1 5060 SIP model only Line 2 5061 Message Waiting Account Null SIP model only I
62. ing conditions and restrictions This manual contains proprietary information belonging to Alvarion Ltd Such information is supplied solely for the purpose of assisting properly authorized users of the respective Alvarion products No part of its contents may be used for any other purpose disclosed to any person or firm or reproduced by any means electronic and mechanical without the express prior written permission of Alvarion Ltd The text and graphics are for the purpose of illustration and reference only The specifications on which they are based are subject to change without notice The software described in this document is furnished under a license The software may be used or copied only in accordance with the terms of that license Information in this document is subject to change without notice Corporate and individual names and data used in examples herein are fictitious unless otherwise noted Alvarion Ltd reserves the right to alter the equipment specifications and descriptions in this publication without prior notice No part of this publication shall be deemed to be part of any contract or warranty unless specifically incorporated by reference into such contract or warranty The information contained herein is merely descriptive in nature and does not constitute an offer for the sale of the product described herein Any changes or modifications of equipment including opening of the equipment not expressly approved
63. ing sequence received up to this time as a complete telephone number This is necessary since the whole telephone number is sent at once and not digit by digit E xx Any number of digits may be dialed A decision that dialing was completed will be reached once is pressed The combination of both schemes means that dialing is completed either after a timeout of T seconds or after pressing Dial timeout The timeout in seconds for the dial timeout dialplan The number of seconds that the unit waits before it sends a complete telephone number This is necessary since the whole telephone number is sent at once and not digit by digit The range is 1 to 60 seconds Default value is 4 seconds Use Use as a quick dial function To send the along with the number to the server uncheck the box The default is enabled RTP Port Range The start and end UDP port range for RTP protocol SIP model onl Recommended values for Start and End ports are in the range nec 1030 65535 The default Start port is 8000 The default End port is 8015 Telephone line Switch the telephone line On or Off The default is Off Voice Gateways System Manual Chapter 3 Using the Web Configuration Server Table 3 6 SIP Configuration H323 Telephone Page Parameters Parameter Description HA mode The High Availability mode defines the support of a secondary Gate Keeper SIP Server for high system availability redundancy an
64. ing the unit 3 9 4 Service Access Configuration Page Home Localization SNMP Service Access RTP Stats Service Access Configuration Select which interfaces are allowed access to the services listed below LAN WAN HTTP Web access V Vv SNMP IV Vv Save Service Access Settings Figure 3 24 Service Access Configuration Page The Service Access Configuration page enables to enable disable access to various services Access from each of the ports LAN or WAN using HTTP and or SNMP can be either enabled or disabled The default for all options is enabled checked Click on the Save Service Access Settings button before leaving the page to save the new settings The new settings will be applied after restarting the unit 66 Operation System Menu 3 9 5 RTP Statistics Page The RTP Stats page enables to monitor from remote the performance of the last call The displayed information includes Bandwidth kb s jitter packet loss and latency Ea The bandwidth relates to the payload only Localization SNMP Service Access RTP Stats Last Call RTP Statistics Packet Lost 00 Jitterfms n Latencyims o Bandwidth kb s 0 Figure 3 25 RTP Statistics Page Voice Gateways System Manual 67 Chapter 3 Using the Web Configuration Server 3 10 Upgrade Page The Upgrade page enables to control the process of downloading either a software file with the extension ro or a configurati
65. isaces 81 A 2 3 Call Alteration ai stn veetcetde cceseclany thescendadatcedccd deqetcaannveieectvanayideaasece nn tte wileadiadenemeces aces 81 A2 A I ea cans oe ste ene nee eee 81 A 25 3 Party Conference 1 cesciccscceccscccssececesccenescagsicensesccnsecaiesncesvscsncdendessteenantseecenseens 82 A 2 6 3 Party Conference 2 x c cc2sedeiiasedswedi ssdia sete ec abe ged ec vidoead esis teers celeste ee 82 A 2 7 Call Waiting Indication Tone 00 00 eceeceeeeeeeeeeeeeeeneeeeeeeeeeeeeeeeeeeeeeseaaeeeseeenaeeesenees 82 A2 8 IN OTN A Nees e et ete eee he a ea 83 Appendix B Default Telephony Parameters Glossary Voice Gateways System Manual xiii Chapter 1 System Description In This Chapter E Introducing the Voice Gateway on page 1 2 E Specifications on page 1 3 Chapter 1 System Description l l Introducing the Voice Gateway Alvarion s Voice Gateway enables operators and service providers using Alvarion s Broadband Wireless Access system to provide subscribers with a number of broadband services transparently The Voice Gateway enables bundling services such as telephony Voice over IP and high speed Internet to end users IP telephony services are supported for standard analog phones or G3 fax machines The VG 1D1V has a single POTS interface and the VG 1D2V has two POTS interfaces The Voice Gateways are available with either H 323 or SIP standard and support both narrow compressed and wide band unc
66. it 3 0x08 TFTP file not defined Voice Gateways System Manual 69 Chapter 3 Using the Web Configuration Server E bit 4 0x10 TFTP session failed E bit 5 0x20 CRC error in downloaded image E bit 6 0x40 incompatible image 3 10 1 1 Examples E An attempt to download from a non existing TFTP server results in code 0x7 0x07 bit 2 0x04 TFTP server not defined plus bit 1 0x02 failed to download new image plus bit O 0x01 upgrade requested or main application not valid E An attempt to download a non existing file results in code Oxb OxOb bit 3 0x08 TFTP file not defined plus bit 1 0x02 failed to download new image plus bit O Ox01 upgrade requested or main application not valid E A successful download results in code 0x01 E A restart without download of main application results in 0x00 70 Operation Restart Page 3 11 Restart Page When settings have been inserted or altered the Voice Gateway must be restarted in order to apply the new settings a ne WAN Restart Warning Restarting the system will terminate all network connections and reset your browser connection Restart Upgrade Restart Logout Figure 3 27 Restart Page Click the Restart button to restart the Voice Gateway Voice Gateways System Manual Chapter 3 Using the Web Configuration Server 3 12 Logout Page Use this page to log out the system Logout Are you sure you
67. k to the VLAN Tagging page In the VLAN page click Add VLAN to open the VLAN Editor In the VLAN Editor enter the follwing for untagged data VLAN ID 90 an arbitrary selection a VLAN ID is required for defining the untagged data This VLAN tag is only used internally in the unit VLAN NAME Untagged VLAN Priority O WAN Yes LAN Yes Click OK to add the VLAN to the Tagged Port Membership table Enter the VLAN ID for untagged data 90 in the fields Untagged VLAN ID on LAN port and Untagged VLAN ID on WAN port and click Save 10 Restart the unit to apply the changes Method 2 1 2 In the VLAN page click Add VLAN to open the VLAN Editor In the VLAN Editor enter the follwing for Voice and Management VLAN Operation VLAN Tagging Menu VLAN ID 60 VLAN NAME Voice amp Mng VLAN Priority 6 WAN Yes LAN No 3 Click OK to add the VLAN to the Tagged Port Membership table 4 Enter the VLAN ID for Voice and Management 60 in the field Default VLAN ID on WAN port and click Save 5 In the Page Selection bar click on VoIP VLAN Configuration to open the VoIP VLAN Configuration page Enter 60 in the VLAN Tag fields for both Call Signaling and RTP Enter 6 in the Priority Tag field for both Call Signaling and RTP Click Save VoIP VLAN Settings Go back to the VLAN Tagging page 6 There is no need to define VLAN in the Port Tag Membership table or in the Untagged WAN and LAN fie
68. l SIP 48 Operation Telephone Menu Table 3 9 SIP Extensions Page Parameters Parameter Description Encode SIP URI with user parameters User Phone will be inserted in the Contact field of SIP uniform resource identifier URI Encode default port in SIP URI Include default port in SIP uniform resource identifier URI even though it is not mandatory according to standard Include default port in INVITE Include default port in the INVITE even though it is not mandatory according to standard Send INVITE with timer header value If the called user agents UA or the SIP Proxy Server SPS requires a session timer for a requested session and the calling UA does not include the Session Expires header in the INVITE message then the called UA or the SPS may reject the request with a 487 request failure message If the use of a session timer is desirable but optional for the session and the calling UA does not include the Session Expires header in the INVITE then the called UA or SPS may add a Session Expires header to the next session setup message In this case the SPS shall add the Session Expires header to the INVITE message and the called UA shall add the Session Expires header to the 200 OK response message The range for the timer header value is from 1 to 999 SIP Session timer value The SIP Session Timer Support feature adds the capability to periodically refresh
69. lds Untagged packets will pass through LAN to WAN and WAN to LAN 7 Restart the unit to apply the changes Voice Gateways System Manual 37 Chapter 3 Using the Web Configuration Server ar Telephone Menu In the SIP model the Telephone page selection bar includes the following options E SIP Section 3 7 1 E SIP Extensions Section 3 7 2 E NAT Section 3 7 3 HM STUN Client Section 3 7 4 E ToS Section 3 7 5 In the H323 model the Telephone page selection bar includes the following options E H323 Section 3 7 1 E ToS Section 3 7 5 38 Operation Telephone Menu Jdid SIP Configuration page SIP SIP Configuration STUN Client ToS SIP H323 Configuration Page Dialplan x t ex T Dial Timeout seconds a Vv Use as a quick dial function RTP Port Range start 8000 EndJ8015 Line 1 Line 2 Telephone Line Con of Con of HA Mode Fixed Fixed Auto C Auto ot ot SIP Server IP primary SIP Server Port primary SIP Server IP secondary SIP Server Port secondary a User Name Password Outgoing Display Name Telephone Number Telephone Domain Name Port MessageWaiting Account Incoming CLIP Keepalive timeout seconds Ring Signal 0 9 Transport Preferred Codecs Set Codecs Fax save Cancel m 6729 GT11A G711U T38 m 6729 ST11A o7
70. m invoice date the Warranty Period During the Warranty Period Alvarion may release to its Customers firmware updates which include additional performance improvements and or bug fixes upon availability the Warranty Bug fixes temporary patches and or workarounds may be supplied as Firmware updates Additional hardware if required to install or use Firmware updates must be purchased by the Customer Alvarion will be obligated to support solely the two 2 most recent Software major releases ALVARION SHALL NOT BE LIABLE UNDER THIS WARRANTY IF ITS TESTING AND EXAMINATION DISCLOSE THAT THE ALLEGED DEFECT IN THE PRODUCT DOES NOT EXIST OR WAS CAUSED BY PURCHASER S OR ANY THIRD PERSON S MISUSE NEGLIGENCE IMPROPER INSTALLATION OR IMPROPER TESTING UNAUTHORIZED ATTEMPTS TO REPAIR OR ANY OTHER CAUSE BEYOND THE RANGE OF THE INTENDED USE OR BY ACCIDENT FIRE LIGHTNING OR OTHER HAZARD Disclaimer a The Software is sold on an AS IS basis Alvarion its affiliates or its licensors MAKE NO WARRANTIES WHATSOEVER WHETHER EXPRESS OR IMPLIED WITH RESPECT TO THE SOFTWARE AND THE ACCOMPANYING DOCUMENTATION ALVARION SPECIFICALLY DISCLAIMS ALL IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE AND NON INFRINGEMENT WITH RESPECT TO THE SOFTWARE UNITS OF PRODUCT INCLUDING ALL THE SOFTWARE DELIVERED TO PURCHASER HEREUNDER ARE NOT FAULT TOLERANT AND ARE NOT DESIGNED MANUFACTURED OR INTENDED FOR USE OR RESALE IN A
71. m an external power supply 100 240 VAC 50 60 Hz 2A max Power Consumption 10 5 W max 1 2 8 Connectors Table 1 8 Connectors Connection Description LAN Type 10 100Base TX RJ 45 Ethernet connection MDI MDIX Cable Length max 100 m PHONE Type RJ 11 1 2 in VG 1D2V Number of Phones REN Up to 5 Cable Length Max 500 m Voice Gateways System Manual Chapter 1 System Description Table 1 8 Connectors Connection Description WAN Type 10 100Base TX RJ 45 Ethernet Connection to SU IDU hub Straight Cable Length max 100 m 12 VDC Standard DC power jack to external power supply 1 2 9 Regulatory Standards Compliance Table 1 9 Standards Compliance Type Standard EMC Low Voltage Directive LVD 73 23 EEC Electromagnetic Compatibility Directive EMC 89 336 EEG Safety IEC 60950 CSA C22 2 No 950 95 UL 1950 AS NZS 3260 Emission Immunity EN 55024 1998 EN 55022 1998 Class B EN 61000 3 2 1995 Harmonics EN 61000 3 3 1995 Flicker FCC part 15 1998 Class B AS NZS 3548 1995 1 2 10 Environmental Table 1 10 Environmental Specifications Item Details Operating temperature OoCto500C Operating humidity 10 95 RH non condensing System Description Chapter 2 Installation In This
72. me The Domain Name for client resolution A string of up to 25 characters The default is null an empty string Click on the Save WAN Settings button before leaving the page to save the new settings The new settings will be applied after restarting the unit Voice Gateways System Manual Chapter 3 Using the Web Configuration Server 3 6 VLAN Tagging Menu The VLAN Tagging page selection bar includes the following options M VLAN Tagging Section 3 6 1 E VoIP VLAN Configuration Section 3 6 3 3 6 1 VLAN Tagging Page The Voice Gateway supports 802 1Q VLAN standard allowing IEEE 802 Local Area Networks LANs of all types to be connected together with Media Access Control MAC Bridges as specified in ISO IEC 15802 3 This standard defines the operation of Virtual LAN VLAN Bridges that permit the definition operation and administration of Virtual LAN topologies within a bridged LAN infrastructure Home VLAN Tagging VolP VLAN Configuration ae VLAN Tagging Tagged Port Membership MLAN ID Priority WAN LAN Name Untagged VLAN ID 1 4094 m m Default VLAN ID 1 4094 o Figure 3 5 VLAN Tagging Page 26 Operation VLAN Tagging Menu The VLAN page enables defining up to 16 VLANs and it includes the following components Table 3 4 VLAN Page Parameters Parameter Description Tagged Port Membership A table displaying the defined VLANs For details on modifyin
73. mpletion means allowing user to skip the timer period T after finished dialing by ending number sequence with no other character is valid for this feature A valid dialplan to accomplish this would be xx xx T Hotline The hotline function allows a predetermined number to be called automatically by waiting T seconds which can also be configured without pressing any keys The hotline function can also be used to receive tones from the Local Exchange This is achieved by leaving the number in the hotline dialplan empty Additional modifications may be required in which case contact Customer Support for assistance The hotline feature is activated by specifying T time out in the dialplan by default T is set to 4 seconds For example xx xx T lt 1234 gt T The number 1234 will be dialed after T seconds Adding Removing Prefixes For outgoing calls VG can add a pre defined prefix to a dialed number via the dialplan and send the number with the added prefix to the server The prefix can automatically replace a dialed digit using the following notation lt dialed substring transmitted string gt For example Set the dialplan to lt 8 1860 gt xxx When dialing 8123 the digit 8 is replaced with 1860 and the actual number sent is 1860123 Operation Telephone Menu In ethereal trace the To field in SIP INVITE is 1860123 For incoming calls VG can remove a prefix in the dialplan
74. natiysczeasnneariecteancaseaae 29 3 6 4 VLAN Configuration Example 1 sicisociccasssccetccnsertceestentincned gnedevacoucnearvabeecesscabeuers 30 3 6 5 VLAN Configuration Example 2 tcc besicnscderekied tied Rdeedeiaas 32 3 6 6 WLAN Configuration Example 3 ccccceceeeeeeeeeeeeeeeeeeeeeaaeeeeeeeaeeeeeeeneeeeestnaaeeees 35 3 Telephone AGU sssissasicsecncicsnacenanescencccedienande nice sees cduadaasnsensnaiiandestacestass sonaienadansaegssueneitasie 38 3 7 1 SIP H323 Configuration Page sii cccd desccdessecorptmmssdicdanicesensedenedansiecartecssteesbioncieens 39 3 7 2 SIP Extensions Page neesosnnseoeerneeeeterrstttresterrnrerrnnrtenntrtttresortnoerennnstnee nenne 48 3 7 3 NAT Traversal Configuration Page SIP Only ssssssssseeeessnesssrrrsssrrnneersnnensnne 50 3 7 4 STUN Client Configuration Page SIP Only 0 ceccceeesesneeeeeeeeeeeeeeeeeeeeetsaeeees 51 3 70 TOS Pagesa cance cacy vaca eE ea EAE e aE aaa o EE E AAEE REECE 52 3 7 6 Line Configuration PAS haat pac as ted elite a dese wesc ante ec eeeoeetaes 53 xii Voice Gateways System Manual Contents 3 7 7 Dial Plan SGNGMOS sx ceasnccors cceasdceosescasdnceavsencnnntas seceananedennchavsseanesdesuneaanduactenceandiees 54 3 8 BW Reservation DRAP Configuration Page cccsssseeceesssceeeesseeeeesneeeeeeensenneeeees 58 3 9 System Men acca sac ake el 62 3 9 1 Set Security Password PACS sei esis sccrccatlesdsiiel vinci sadaedadealgdneusriuiauiieamiediedils 62
75. ncoming CLIP On Off Off Keepalive timeout Ring signal 10 65535 seconds 0 9 1200 seconds 0 Transport SIP model only UDP TCP UDP Codecs and Fax Configuration Jitter Buffer Adaptive Jitter Buffer 100 300 milliseconds 100 milliseconds Fixed Jitter Buffer 100 300 milliseconds 40 milliseconds Automatically switch to Fixed Jitter Yes checked No Unchecked Buffer unchecked Codec G711A Select Deselect Yes checked No unchecked Yes checked SS Not applicable EC Enable Disable Enabled Packet 10 150 milliseconds 20 ms Keypad SIP None RFC2833 None SIP INFO H323 H225 H245 RFC2833 None Priority 1 3 1 Voice Gateways System Manual Chapter 3 Using the Web Configuration Server Table 3 18 Parameters Summary Parameter Range Options Default G711U Select Deselect Yes checked No unchecked Yes checked SS Not applicable EC Enable Disable Enabled Packet 10 150 milliseconds 20 ms Keypad SIP None RFC2833 None SIP INFO H323 H225 H245 RFC2833 None Priority 1 3 2 G729 Select Deselect Yes checked No unchecked Yes checked SS Enable Disable Disabled EC Enable Disable Enabled Packet 10 150 milliseconds 30 ms Keypad SIP None RFC2833 SIP RFC2833 SIP INFO H323 H225 H323 H225 H245 RFC2833 None Priorit
76. nnccannaccenncnuetenanaceen 2 sted ie le L sisisi nanasan nananana sannana eeeeeeeeeneeereereee 3 1 2 1 Telephony and Fax ServiCes cs ccccceeecceeeseseeeeeeeseeceeeeseneeeenseneeeeseseeeeeseeeeneees 3 IAL A E E T 3 LAS Voice UCIT Aee 4 1 2 4 Configuration and Management c cccessssceeesseneceeeeeeeeeeeeseeeeeeeeseeneneeeeeeeees 4 1 2 5 Bridge ICTIONNALILY 2 cccccceseceeceesecceeeeseeeceeeesneeeeeeeaeeeseeeeceeensneeeeessecneeseeeeneees 4 126 IIE eE EEEE 5 laz Electrical cE 5 1 2 2 f CONNECTE 5 1 2 9 Regulatory Standards Compliance ccccesseseceeeeceeeeeeeeeeeeeeeeeeeneeeeeeeteeeneaes 6 2O Env ONION e EEEE 6 Chapter 2 Installation 2 1 Installation Requirements snnsssnunnnennnnnunnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnn nunnana nnmnnn 8 24A Packing eee ee ene ore ne ee ee eer 8 2 1 2 Additional Installation Requirement ccccccceesseeeenceceeeeeeeeeeeeeeeeeeeeneeeeeneeees 8 2 2 Front and Rear Panel Components ccccseeeeeseeeeeeeeseneeeeeeeeeeeeeneeeeeeeesneeeeeseeeeneneeneees 9 22l TONNE CIOS enirere cede SL A rO 9 2 2 2 Reset to Factory Default Configuration ssssssseessneessserrneerrnrrrrneersrtrrnserrnnnrrrne 9 223 LEDS ei a E a a nee ee 10 2 3 Installation and COMMISSIONING ssssunsssunnennnnnnnennnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnn nnne 11 2 4 Notes on Using the Voice Gateways in Alvarion s SystemsS ccsssseeses
77. ntil an actual call is established Once a call is established the unit will use a packet size according to the actual value being used for the call The available range is 10 to 100 milliseconds in multiples of 10 10 20 100 The default value is 30 milliseconds 60 Operation BW Reservation DRAP Configuration Page Table 3 13 DRAP Configuration Page Parameters Parameter Description Vocoder Type The Vocoder Type parameter defines the codec to be used until an actual call is established It is recommended to set a value that supports the worst case scenario e g the codec with the highest bandwidth requirement Typically G711 should be configured except in networks where only G729 is used NOTE The configured Vocoder Type is used by the unit until an actual call is established Once a call is established the unit will use the actual codec type being used for the call The available options are G711 and G729 The default is G729 Click on the Save DRAP Settings button before leaving the page to save the new settings The new settings will be applied after restarting the unit Voice Gateways System Manual Chapter 3 Using the Web Configuration Server 3 9 System Menu The System page selection bar includes the following options E Security Section 3 9 1 E Localization Section 3 9 2 E SNMP Section 3 9 3 E Service Access Section 3 9 4 E RTP Stat
78. nu 3 9 3 SNMP Configuration Page Localization SNMP S RTP Stats SNMP Configuration SNMP Trap Configuration Trap Destination 1 Trap Destination 2 Trap Destination 3 Trap Destination 4 Trap Destination 5 rr Trap Destination 6 FT SNMP MIB Parameter Configuration Read Community public Write Community private Save SNMP Settings Logout Figure 3 23 SNMP Configuration Page The SNMP Configuration page includes the following components Table 3 16 SNMP Configuration Page Parameters SNMP Trap Configuration Trap Destination 1 to Trap Specify up to 6 IP addresses to which SNMP traps should be sent destination 6 Only these stations will be able to manage the Voice Gateway If all Trap Destinations are null SNMP traps will be sent as broadcasts and any station will be able to manage the Voice Gateway SNMP MIB Parameter Configuration Voice Gateways System Manual 65 Chapter 3 Using the Web Configuration Server Table 3 16 SNMP Configuration Page Parameters Read Community The read community string up to 20 printable characters case sensitive Default string is public Write Community The write community string up to 20 printable characters case sensitive Default string is private Click on the Save SNMP Settings button before leaving the page to save the new settings The new settings will be applied after restart
79. ompressed speech codecs silence suppression with comfort noise line echo cancellation and regional telephone parameters Class 5 services such call waiting and 3 party conference call are also supported Up to 3 telephones can be connected in series to each telephone port Daisy chaining of Voice Gateways enables the service provider to offer certain end users for example small offices additional telephone numbers The Voice Gateway also supports Internet access or any other Ethernet based services The unit can be installed behind a router NAT due to NAT traversal support allowing signaling as well as voice packets to correctly reach Softswitch or Gatekeeper for bi directional call initiations The Gateway can handle up to 16 simultaneous VLANs enabling the operator to offer different services to different end users behind the unit These Gateways incorporate the proprietary DRAP Dynamic Resources Allocation Protocol protocol for automatic registration and allocation of resource DRAP isa protocol based on IP UDP between the Gateway and a DRAP server e g the BreezeMAX base station The protocol provides an auto discovery mechanism for the Gateway so no specific configuration is required and the Gateway can automatically locate and register with the DRAP server The protocol uses a few simple messages enabling a Voice Gateway to request resources when calls are made and the DRAP server to dynamically allocate them The Voice Gateways
80. on file with the extension ini from a TFTP server or from an HTTP server sbi pure Upgrade Warning The upgrade process will reset the unit into the download mode This will terminate all network connections and reset your browser connection system X Upgrade Type TFTP Upgrade Restart Host fisz 168 1 11 Filename fiocal_ringing_off ini Start TFTP Upgrade Figure 3 26 Upgrade Page The Upgrade page includes the following components Table 3 17 Upgrade Page Parameters Upgrade Type Auto The Voice Gateway will automatically select the server type for download TFTP Downloads the file from the TFTP server according to the specified host address HTTP Downloads the file from the HTTP server according to the specified URL Not implemented in the current release Host TFTP The IP address of the TFTP server URL HTTP Auto Use the following syntax IP filename E g 192 168 254 1 DMA0027R2F201 ro File name The file name in the HTTP TFTP server of the software or the configuration ini file Up to 25 characters Click on the Start Auto HTTP TFTP Upgrade button to start the download process The downloading and installation of the new SW version or configuration 68 Operation Upgrade Page file is done automatically including a restart of the unit When the installation is complete and the unit has restarted the Home Product Info page will be displayed if not cli
81. parameters are configured manually IP Address The IP address of the unit MAC Address Subnet Mask The MAC address of the unit The IP Subnet Mask Default Gateway The Default Gateway address DNS Address IP DNS Server address Domain Name The Domain Name as defined in the WAN Configuration page VLAN Tag The VLAN ID tag defined for management traffic Priority Tag The Priority tag defined for management traffic Click on the Update button to refresh the display Voice Gateways System Manual Chapter 3 Using the Web Configuration Server 3 5 2 WAN Configuration Page WAN Configuration WAN Configuration Device Operating Mode Bridge Obtain WAN configuration using DHCP Client identity Logout O Standard C Custom Vendor ID C Specify static WAN configuration IP Address 192 168 254 254 Subnet Mask 255 255 255 0 Default Gateway a DNSAddres Hostname TO Domain Name Save WAN Settings Figure 3 4 WAN Configuration Page The WAN Configuration page includes the following components Table 3 3 WAN Configuration Page Parameters Device Operating Mode The operating mode of the unit In current version the operation mode is always Bridge Obtain WAN configuration using Select this option to obtain IP parameters from a DHCP server DHCP See also Section 2 2 2 24 Operation WAN Menu
82. privileges the default user name is operator and the default password is installer To login with administrator privileges partial access and read write privileges the default user name is admin No password is required 4 The Web Configuration Server main view appears on the screen Voice Gateways System Manual 17 Chapter 3 Using the Web Configuration Server wun Using the Web Configuration Server The Web Configuration Server view consists of a number of menu links to the left Clicking on each of them will display the configuration status page for the selected menu item with the applicable content configurable parameters options or status information in the main area Several pages include a page selection bar at the top of the page enabling selection between several pages related to the same menu item The displayed pages may vary depending on user privileges Security Localization SNMP Service Access RTP Stats SNMP Configuration Page Selection Bar vatior SNMP Trap Configuration System Trap Destination 1 Upgrade Trap Destination 2 Restart Trap Destination 3 Logout Trap Destination 4 Trap Destination 5 Trap Destination 6 l SNMP MIB Parameter Configuration Read Community public Write Community private Page Main Area Save SNMP Settings Figure 3 1 Web Configuration Page A Many pages include a Save Settings button Click on the Save Settings button before selecting
83. r products and or services referenced here in are either registered trademarks trademarks or service marks of Alvarion Ltd All other names are or may be the trademarks of their respective owners Statement of Conditions The information contained in this manual is subject to change without notice Alvarion Ltd shall not be liable for errors contained herein or for incidental or consequential damages in connection with the furnishing performance or use of this manual or equipment supplied with it Warranties and Disclaimers All Alvarion Ltd Alvarion products purchased from Alvarion or through any of Alvarion s authorized resellers are subject to the following warranty and product liability terms and conditions Exclusive Warranty a Alvarion warrants that the Product hardware it supplies and the tangible media on which any software is installed under normal use and conditions will be free from significant defects in materials and workmanship for a period of fourteen 14 months from the date of shipment of a given Product to Purchaser the Warranty Period Alvarion will at its sole option and as Purchaser s sole remedy repair or replace any defective Product in accordance with Alvarion standard R amp R procedure iv Voice Gateways System Manual Legal Rights b With respect to the Firmware Alvarion warrants the correct functionality according to the attached documentation for a period of fourteen 14 month fro
84. r to a SIP proxy server even though the Voice Gateway is connected behind a NAT device Port forwarding may need to be activated for all telephone ports used by Voice Gateway For example RTP port range and SIP signaling ports The Keep alive timeout parameter in the Telephony page may also need to be set to a value lower than 1200 seconds to ensure that the Voice Gateway will not lose registration to the SIP server Home STUN Client ToS DiffServ Line Configuration Static NAT Traversal Configuration External NAT mapped IP Address l Static NAT Mode C on Use above configured IP address as mapped IP address G Auto Automatic detect and set mapped IP address Restart C on Logout Figure 3 16 NAT Traversal Configuration Extensions Page The NAT Traversal Configuration page includes the following components Table 3 10 NAT Traversal Configuration Page Parameters External NAT mapped The IP address that the NAT device uses on the WAN side If the Voice IP Address Gateway is set to Auto NAT mode see below the IP address of the outside IP will be automatically inserted If the NAT Mode is set to On a NAT IP Address must be set 50 Operation Telephone Menu Table 3 10 NAT Traversal Configuration Page Parameters Parameter Description Static NAT Mode The NAT mode E On Enable NAT Traversal function using manual setting E Auto Enter NAT mode if any of the following con
85. ress The MAC address of the unit Serial Number The serial number of the unit Product number Not Used Product revision The hardware revision Production week Production date in the format lt yy gt w lt wws gt lt yy gt is the year two last digits and ww is the week two digits Default configuration The unit s configuration Downloader revision The revision of the SW download SW module 20 Operation Home Menu Product info Page Table 3 1 Product Info Page Parameters Parameter Description Reported download status The status of the SW download operation For more details refer to Section 3 10 1 Main software revision The unit s main SW version Operator defaults revision The custom ini file if exists In any case of contact with Alvarion Customer Service include the Default configuration Downloader revision Main software revision and Operator defaults revision ini file if exists Voice Gateways System Manual 21 Chapter 3 Using the Web Configuration Server 3 5 WAN Menu The WAN menu page includes settings related to the operation and functionality on the WAN network side of the unit Ea Be careful when setting these parameters to avoid conflicts in the network The WAN page selection bar includes the following options E WAN Status Section 3 5 1 E WAN Configuration Section 3 5 2
86. s Section 3 9 5 3 9 1 Set Security Password Page Security Localization Change security settings 0 User name ae System a Up de Old password Do New pasaword Confirm new password Save Password Access x Save Access Mode Figure 3 21 Set Security Password Page The Set Security Password page includes the following components 62 Operation System Menu Table 3 14 Set Security Password Page Parameters Parameter Description User name Enter the user name The user name for users with operator privileges full access and read write privileges is operator and for user with administrator privileges partial access and read write privileges is admin These user names cannot be changed Old password A password used previously The default password for users with operator privileges is installer No password is required for users with administrator privileges New password Enter the new password A password includes up to 20 printable characters and is case sensitive A null empty string means no password Confirm new password Enter the new password again must be the same as above Access Select the mode in which the PC can manage the IDU DV unit The PC can manage the unit through the LAN User Ethernet port the WAN Radio or BOTH It is recommended that you select BOTH Click on the Save Password button before leaving the page to save the new pas
87. standard 1991 that uses TCM Trellis Coded Modulation a technique for forward error correction modulation at 12 000 and 14 400 bps for Group 3 faxes It adds TCM to the V 29 standard at 7 200 and 9 600 bps to allow transmission over noisier lines It also defines special functions echo protection turn off sequences etc for half duplex operation Modulation use is a half duplex version of V 32bis An ITU standard 1976 for synchronous 4 800 7 200 and 9 600 bps full duplex modems using QAM Quadrature Amplitude Modulation on four wire leased lines It has been adapted for Group 3 fax transmission over dial up lines at 9 600 and 7 200 bps An ITU standard 1991 for asynchronous and synchronous 4 800 7 200 9 600 12 000 and 14 400 bps full duplex modems using TCM and echo cancellation It supports rate renegotiation which allows modems to change speeds as required Virtual Local Area Network A group of devices on one or more LANs that are configured with the same VLAN ID so that they can communicate as if they were attached to the same wire when in fact they are located on a number of different LAN segments Used also to create separation between different user groups Voice over Internet Protocol Provides an advanced digital communications network that bypasses the traditional public switched telephone system and uses the Internet to transmit voice communication VoIP enables people to use the Internet as the transmission me
88. sword Click on the Save Access Mode button before leaving the page to save the access mode The new settings will be applied after restarting the unit When upgrading the unit the new password is retained and does not revert to the default password Voice Gateways System Manual 63 Chapter 3 Using the Web Configuration Server 3 9 2 Localization Page Home Security Localization SNMP Seri ss RTP Stats Localization NTP Server Time Zone cmr 01 00 7 M Adjust clock for daylight savings Restart Logout T Save Localization Settings Figure 3 22 Localization Page The Localization page includes the following components Table 3 15 Localization Page Parameters NTP Server The IP address of the NTP server optional If an IP address is configured the NTP server usage is activated The feature must be activated to support FSK based caller ID The default is disabled no IP address Time Zone The appropriate time zone Use the drop down list to change the time zone Adjust clock for daylight By checking the Adjust clock to daylight savings the Voice Gateway savings will automatically adjust to daylight saving time set the time one hour ahead The default is enabled checked Click on the Save Localization Settings button before leaving the page to save the new settings The new settings will be applied after restarting the unit 64 Operation System Me
89. t Membership parameters VLAN page and VoIP VLAN parameters VoIP VLAN Configuration page Telephony parameters per line in the SIP Configuration H323 Telephone page Telephone Line Enable Disable primary SIP Server H323 Gate Keeper parameters User Name and Password SIP model Telephone Number Telephone domain name SIP model Certain H323 Gatekeepers require configuration of a unique H323 Alias WAN IP parameters WAN Configuration page For operation as a DHCP client check the Obtain WAN Configuration dynamically For static IP configuration check the Specify fixed WAN configuration option and specify the IP Address Subnet Mask and Default Gateway Restart the unit from the Restart page If VLANs are configured for management you will lose management from the PC unless the packets are tagged from the PC towards the Voice Gateway To resume management capabilities return to factory defaults see Section 2 2 2 Disconnect the PC used for configuration Use a straight Ethernet cable to connect the WAN port on the rear panel of the unit to the Ethernet port of the SU IDU The length of the indoor to outdoor Ethernet cable should not exceed 90 meters The length of the Ethernet cable connecting the indoor unit to the user s equipment together with the length of the Indoor to Outdoor cable should not exceed 100 meters Connect the data equipment using a 10 100 Base T Ethernet cable to the LAN port The length of the
90. tagged VLAN Priority O WAN Yes LAN Yes 8 Click OK to add the VLAN to the Tagged Port Membership table 9 Enter the VLAN ID for untagged data 300 in the fields Untagged VLAN ID on LAN port and Untagged VLAN ID on WAN port and click Save 10 Restart the unit to apply the changes 3 6 6 VLAN Configuration Example 3 This example describes how to define the following configuration M One Voice Gateway E VLAN ID 60 VLAN Priority 6 for Voice call signaling RTP and RTCP and Management packets on the WAN port E No VLAN for data packets on WAN and LAN ports No VLAN VLAN 60 Figure 3 11 VLAN Configuration Example 3 Voice Gateways System Manual 35 Chapter 3 Using the Web Configuration Server 3 6 6 1 3 6 6 2 36 Method 1 1 2 In the VLAN page click Add VLAN to open the VLAN Editor In the VLAN Editor enter the follwing for Voice and Management VLAN VLAN ID 60 VLAN NAME Voice amp Mng VLAN Priority 6 WAN Yes LAN No Click OK to add the VLAN to the Tagged Port Membership table Enter the VLAN ID for Voice and Management 60 in the field Default VLAN ID on WAN port and click Save In the Page Selection bar click on VoIP VLAN Configuration to open the VoIP VLAN Configuration page Enter 60 in the VLAN Tag fields for both Call Signaling and RTP Enter 6 in the Priority Tag field for both Call Signaling and RTP Click Save VoIP VLAN Settings Go bac
91. tion Server Table 3 13 DRAP Configuration Page Parameters Parameter Description Acknowledge Time The Acknowledge Time is the timeout out to be used between allocation requests If no confirmation is received within this time a new allocation request should be sent The range is 1 to 10 x 100 milliseconds The default is 3 300 milliseconds Clear Count The Clear Count parameter indicates the number of allocation requests ALLOC that can be sent without being acknowledged before clearing all pending reservation attempts Note Established reservations existing calls are not cleared The range is 1 to 10 The default is 2 Retry Count RTP Packing Ratio The Retry Count parameter indicates the number of allocation requests ALLOC that can be sent without being acknowledged before reaching a decision that the unit should search for another server When this number is reached established reservations are to be cleared existing calls are disconnected and auto discovery procedure is initiated The range is 1 to 10 The default is 5 The RTP Packing Ratio parameter defines the packet size to be used until an actual call is established It is recommended to set a value that supports the worst case scenario e g the smallest expected size 20 milliseconds that results in the highest expected number of packets per second NOTE The configured RTP Packing Ratio is used by the unit u
92. ue that merges binary data into a carrier It creates only two changes in frequency one for 0 another for 1 A 64 Kbps PCM voice coding technique Described in the ITU T standard in its G series recommendations A compression technique where voice is coded into 8 Kbps streams There are two variations of this standard G 729 and G 729 Annex A that differ mainly in computational complexity both provide speech quality similar to 32 kbps ADPCM Described in the ITU T standard in its G series recommendations An ITU standard protocol for control signaling in an H 323 audio or video environment H 225 uses messages defined in H 245 to establish the call over the Registration Admission and Signaling RAS channel An ITU standard protocol for control messages in an H 225 audio or videoconferencing call The messages are used for flow control encryption and jitter management as well as for initiating the call negotiating which features should be used and terminating the call It also determines which side is the master for issuing various commands A protocol suite defined by ITU T for voice transmission over internet Voice over IP or VoIP In addition to voice applications H 323 provides mechanisms for video communication and data collaboration in combination with the ITU T T 120 series standards High Availability Voice Gateways System Manual Glossary HTTP IDU IEC IEEE IEEE 802 1p IEEE 802 1Q IE
93. vent R3 telephone number Place on going call on hold dial tone inquire new call to a third party and mix all session into a conference when third party has answered R5 End conference with third party and return to first initiated call session A 2 6 3 Party Conference 2 Description Two on going calls active mix them into a conference session Table A 7 3 Party Conference Service 2 Action Event R3 Start conference with all active parties mix audio streams R5 End conference with third party and return to first initiated call session A 2 7 Call Waiting Indication Tone Description Available only when there are no calls active in progress Table A 8 Call Waiting Indication Tone Service Action Event 43 Enable Call Waiting indication tone 43 Disable Call Waiting indication tone calling party will hear a busy tone when calling 82 Voice Gateways System Manual Using the Class 5 Services Table A 8 Call Waiting Indication Tone Service Action Event 434 Informs about present Call Waiting indication tone configuration Three short beeps off Two long beeps on A 2 8 Call Forward Description Available only when there are no calls active in progress Table A 9 Call Forward Service Action Event 21 lt telephone number gt 21 Enable Call Forward and
94. y 1 3 Not Applicable T38 Fax Enable Disable Enable SIP Extensions Page SIP model only Support PRACK method with Yes No No unchecked provisional response reliability Encode SIP URI with user Yes No No unchecked parameters Encode default port in SIP URI Yes No No unchecked Include default port in INVITE Yes No Yes checked Send INVITE with timer header Yes No plus a value in the No unchecked value range 1 999 seconds if Yes Thede value ieni e default value checked is selected SIP Session timer value Yes No plus a value in the No unchecked range 1 999 seconds if Yes The deut vaus ienai e default value checked is selected 76 Operation Parameters Summary Table 3 18 Parameters Summary Parameter Range Options Default Use NOTIFY message to keep alive Yes checked No 15 seconds the session with SIP proxy every X unchecked seconds 0 99999 seconds NAT Traversal Configuration Page SIP model only External NAT mapped IP Address IP address Null Static NAT Mode On Auto Off Off STUN Client Page STUN Client Mode On Off Off STUN Server Address IP address Domain Null STUN Server Port 1 65534 3478 ToS Page Call signaling Packets 0 255 or null 192 RTP Packets 0 255 or null 160 SNMP Packets 0 255 or null 0 Default setting 0 255 or null 0 Line Configuration Page CLIP Standard A list Sweden DRAP Configuration Page

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