espace U1960 Unified Gateway V100R001C01 Product Overview

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1 V100R001C01 Issue Date HUAWEI TECHNOLOGIES CO., LTD.

2 2012. All rights reserved. No part of this document may be reproduced or transmitted in any form or by any means without prior written consent of Huawei Technologies Co., Ltd. Trademarks and Permissions and other Huawei trademarks are trademarks of Huawei Technologies Co., Ltd. All other trademarks and trade names mentioned in this document are the property of their respective holders. Notice The purchased products, services and features are stipulated by the contract made between Huawei and the customer. All or part of the products, services and features described in this document may not be within the purchase scope or the usage scope. Unless otherwise specified in the contract, all statements, information, and recommendations in this document are provided "AS IS" without warranties, guarantees or representations of any kind, either express or implied. The information in this document is subject to change without notice. Every effort has been made in the preparation of this document to ensure accuracy of the contents, but all statements, information, and recommendations in this document do not constitute a warranty of any kind, express or implied. Huawei Technologies Co., Ltd. Address: Website: Huawei Industrial Base Bantian, Longgang Shenzhen People's Republic of China i

3 Contents Contents 1 Introduction Product Positioning Highlights Application Scenario Single-Node Scenario Distributed Networking Scenario Centralized Networking Scenario Product Architecture Shelf Board Functions and Features Basic Voice Services Supplementary Services Advanced Services Voice Conference Voice Mailbox Services Automatic Switchboard Service CDR Intelligent Routing Security Operation, Maintenance, and Management Web Management System Web Self-Service System LMT CLI Management Mode Technical Specifications Physical Specifications Performance and Capacity Ports and Protocols Standard Compliance ii

4 1 Introduction 1 Introduction About This Chapter This topic describes the positioning and features of the espace U Product Positioning The espace U1900 series are switches used in Huawei IP Telephony (IPT) solutions. The espace U1900 series provide professional voice over IP (VoIP) solutions that meet communication requirements of various enterprises. 1.2 Highlights The espace U1960 provides rich services and ports. It is highly reliable and easy to deploy and maintain. 1.1 Product Positioning The espace U1900 series are switches used in Huawei IP Telephony (IPT) solutions. The espace U1900 series provide professional voice over IP (VoIP) solutions that meet communication requirements of various enterprises. Table 1-1 shows the models and application scenarios of the espace U1900 series. Table 1-1 Models and application scenarios of the espace U1900 series Model espace U1910 espace U1930 espace U1960 espace U1980 Application Scenario Small- and medium-sized enterprises or branches of large-sized enterprises that have less than 100 users. Small- and medium-sized enterprises or branches of large-sized enterprises that have 100 to 300 users. Small- and medium-sized enterprises or branches of large-sized enterprises that have 300 to 1000 users. Large-sized enterprises that have 1000 to users. 1

5 1 Introduction The espace U1960 is a medium-capacity product that can be used as a VoIP communication device in small and medium-sized enterprises with no more than 1000 users or a local access gateway in branches of large-sized enterprises, improving the communication efficiency and reducing costs. Figure 1-1 shows the appearance of espace U1960. Figure 1-1 espace U1960 appearance The espace U1900 series use a highly-integrated SIP softswitch as the core and support both narrowband and broadband services. The espace U1900 series can connect to analog phones and IP phones at the same time: Connect to local analog phones directly. Use IP bearer networks to connect to phones through espace IADs. Use IP bearer networks to connect to IP phones. The espace U1900 series can connect to PSTN networks or dedicated network voice switches through digital, analog and broadband SIP trunks also IP Phones can connect in the network in a central site or remote site. 1.2 Highlights The espace U1960 provides rich services and ports. It is highly reliable and easy to deploy and maintain. Rich Services and Ports Provides high-quality, built-in voice conference resources and web conference management portal for easy use and management. Provides built-in voice mailboxes to implement the voic function and prevent call loss. Supports PRI, SS7, R2 (MFC), QSIG, AT0 and SIP signaling messages and protocols and integrates narrowband and broadband services. 2

6 1 Introduction High Reliability and Low Costs Provides % system reliability and supports both AC and DC power supplies with 1+1 redundancy backup. Supports the local regeneration function at enterprise branches. Adapts to requirements of small- and medium-sized enterprises and reduces the operation and maintenance costs for enterprises. Easy Deployment and Maintenance Can be expanded and upgraded to UC applications, protecting customers' investments. Supports various networking modes, including the single-node system, distributed networking, and centralized networking. Both IPv4 and IPv6 networks are available, allowing for flexible deployment. Supports unified network management, and provides a visual maintenance and operation tool. Provides a built-in web configuration tool for deployment configuration, maintenance, and management. Provides a built-in web management system that allows users to schedule conferences, modify forward-to numbers and ONLY numbers, activate the voice mailbox service. 3

7 2 Application Scenario 2 Application Scenario About This Chapter The espace U1960 is applicable to the following application scenarios. 2.1 Single-Node Scenario In the single-node scenario, an espace U1960 is deployed and connected to the local carrier network. Voice terminals such as IP phones, analog phones, and PC clients connect to the espace U1960 and provide rich voice services for the enterprise. This network applies to small- and medium-sized enterprises with no branches. 2.2 Distributed Networking Scenario In the distributed networking scenario, the espace U1960 is deployed on the central node, and the espace U1930 or espace U1910 is deployed on a branch node. This networking applies to medium-sized enterprises with branches. 2.3 Centralized Networking Scenario In the centralized networking scenario, the espace U1980 is deployed on the central node, and espace U1960s are deployed on branch nodes. This networking applies to large enterprises with branches. 2.1 Single-Node Scenario Typical Network In the single-node scenario, an espace U1960 is deployed and connected to the local carrier network. Voice terminals such as IP phones, analog phones, and PC clients connect to the espace U1960 and provide rich voice services for the enterprise. This network applies to small- and medium-sized enterprises with no branches. Figure 2-1 shows the typical network of the single-node scenario. 4

8 2 Application Scenario Figure 2-1 Typical network of the single-node scenario Network Description Analog phones and fax machines connect to the espace U1960 through analog phone cables. IP phones and SoftPhone connect to the espace U1960 in LAN mode. A maximum of 1000 voice users can register with the espace U1960. The espace U1960 connects to the PSTN through the PRI, SS7, R2 or QSTG trunk. The espace U1960 provides voice mailbox and voice conference services for enterprise users. The espace U1960 provides a built-in web management system for configuring services. A UC server can be added to the network to provide UC, Call Detail Record (CDR), and Element Management System (EMS) services. 2.2 Distributed Networking Scenario Typical Network In the distributed networking scenario, the espace U1960 is deployed on the central node, and the espace U1930 or espace U1910 is deployed on a branch node. This networking applies to medium-sized enterprises with branches. Figure 2-2 shows the distributed networking. 5

9 2 Application Scenario Figure 2-2 Distributed networking Network Description The network is described as follows: A maximum of 200 branch nodes are supported in a distributed network. The central node and branch nodes connect to the PSTN through digital or analog trunks. The central and branch nodes are connected through SIP trunks and use the heartbeat mechanism to monitor the running status of the peer devices. When the central node and branch nodes are connected properly: Users and services are configured and controlled in a unified manner on the central node. SIP phones on branch nodes are directly registered with the central node. Analog phones on branch nodes are registered with the central node through branch node proxies. Branch nodes synchronize SIP user data from the central node. When the central node is faulty or disconnected from branch nodes: The basic call function remains available for analog users on branch nodes. When detecting that the central node is faulty or disconnected, IP phones on a branch node automatically set the branch node as the SIP server. The basic call function remains available for SIP users on the branch node. 6

10 2 Application Scenario 2.3 Centralized Networking Scenario Typical Network In the centralized networking scenario, the espace U1980 is deployed on the central node, and espace U1960s are deployed on branch nodes. This networking applies to large enterprises with branches. Figure 2-3 shows the centralized networking. Figure 2-3 Centralized networking Network Description The network is described as follows: A maximum of 200 branch nodes are supported in a centralized network. The central node and branch nodes connect to the PSTN through digital or analog trunks. The central and branch nodes are connected through SIP trunks and use the heartbeat mechanism to monitor the running status of the peer devices. When the central node and branch nodes are connected properly: Users and services are configured and controlled in a unified manner on the central node. Branch nodes synchronize SIP user data from the central node, and register analog users on branch nodes with the central node. 7

11 2 Application Scenario When the central node is faulty or disconnected from branch nodes: The basic call function remains available for analog users on branch nodes. When detecting that the central node is faulty or disconnected, IP phones on a branch node automatically set the branch node as the SIP server. The basic call function remains available for SIP users on the branch node. 8

12 3 Product Architecture 3 Product Architecture About This Chapter This topic describes the hardware structure of the espace U Shelf A shelf provides a space for placing and connecting espace U1960 internal components. It also protects the components from contamination and external damage. 3.2 Board The espace U1960 has main control boards (SCU), MTU boards, Analog Subscriber Interface (ASI) boards, and FXO/FXS Interface Unit (OSU) boards. 3.1 Shelf Appearance A shelf provides a space for placing and connecting espace U1960 internal components. It also protects the components from contamination and external damage. The espace U1960 adopts the standard 2U (88.9 mm) shelf. The dimensions (H x W x D) of the shelf are 86.1 mm x 442 mm x 310 mm. The self can be installed in a 19-inch cabinet that meets the International Electrotechnical Commission (IEC) standard. Figure 3-1 shows the espace U1960 front panel of. 9

13 3 Product Architecture Figure 3-1 Front panel Slots Slots are located on the front panel of the shelf. The espace U1960 provides one main control board slot, seven interface board slots, two power supply sockets, and one fan tray assembly slot, as shown in Figure 3-2. Slots 0 to 6 are service board slots, which are used to install the MTU, ASI and OSU boards. Slot 7 is the main control board slot, which is used to install a SCU board. The fan tray assembly is vertically mounted on the left side of the device. It has three fans to provide optimal heat dissipation for the system, which ensures device stability. The power module is deployed on the right side of the device in 1+1 backup mode. It supports DC or C power supply. Figure 3-2 Slot distribution The number of espace U1960 interface boards is determined by the system capacity. Filler panels must be inserted into blank slots. 10

14 3 Product Architecture 3.2 Board The espace U1960 has main control boards (SCU), MTU boards, Analog Subscriber Interface (ASI) boards, and FXO/FXS Interface Unit (OSU) boards. Board Panel and Functions CVP SCU A CVP board provides the following functions: Provides the SoftSwitch functions. Processes media control protocols. Supports L2 switching and Time Division Multiplex (TDM) switching. MTU An SCU board provides the following functions: Provides the SoftSwitch functions. Processes media control protocols. Supports three working modes: single-network-port, dual-network-port, and triple-network-port. Supports L2 switching and TDM switching. An MTU board provides media resource functions for voice services and digital trunk ports for connection to peer offices through digital trunks. 11

15 3 Product Architecture Board Panel and Functions ASI OSU An ASI board provides 32 Foreign Exchange Subscriber (FXS) ports. An OSU board provides 12 FXS ports and 12 Foreign Exchange Office (FXO) ports. The OSU board can support local survival in case of power electrical off. 12

16 4 Functions and Features 4 Functions and Features About This Chapter The espace U1960 provides comprehensive voice services and supplementary services for users Basic Voice Services Supplementary Services 4.3 Advanced Services The espace U1960 supports user rights management by level and call barring policies. 4.4 Voice Conference 4.5 Voice Mailbox Services The espace U1960 provides the Voice Mailbox Services (VMS) by default. It can also connect to the Unified Messaging System (UMS) to store and manage voice messages in a unified manner to enable uses to retrieve voice messages at any time anywhere by dialing the voice message retrieving prefix. 4.6 Automatic Switchboard Service 4.7 CDR The espace U1960 generates CDRs for users and write and save the CDRs in a CDR server. Other applications can obtain CDRs from a CDR server to parse them. 4.8 Intelligent Routing The intelligent routing service automatically selects office routes when an IP trunk or a TDM trunk is faulty, and provides routing polices to increase communication reliability and minimize communication costs. 4.9 Security The espace U1960 provides enhanced security measures for users. 13

17 4 Functions and Features 4.1 Basic Voice Services Voice Communication The espace U1960 supports basic voice communication, including: Intra-office communication Intra-office users under the espace U1960 can use various supported terminals to make voice calls to each other. Narrowband trunk-based communication The espace U1960 connects to the PSTN or TDM PBX through digital trunks such as PRI, R2, SS7 (ISUP/TUP), or Q Signaling (QSIG). SS7 is short for signaling system No. 7, ISUP for integrated services digital network user part, and TUP for telephone user part. Users can make voice calls and receive incoming calls based on narrowband trunks. Broadband trunk-based communication The espace U1960 uses the SIP trunk to connect to the IP PBX, softswitch, or IMS. Voice calls can be made between intra-office users and the users under other devices. The espace U1960 can be connected to broadband terminals (including SoftPhone, espace Desktop, and SIP SoftConsole), and access devices (including IADs). Point-to-Point Multimedia Communication The espace U1960 supports the point-to-point (P2P) multimedia communication service, allowing SIP-based multimedia terminals to communicate with each other. Supports the espace Desktop and the espace Supports P2P video calls between intra-office users and SIP-based P2P video calls between intra-office and outer-office users. Call Rights Control The espace U1960 allows users to make intra-office calls, local calls, national toll calls, and international toll calls. In addition, a maximum of 32 types of call rights can be defined to restrict users from making specified types of calls. Number Analysis and Processing The espace U1960 analyzes the calling and called numbers based on their prefix and length to control incoming and outgoing call rights. The calling number is analyzed before the called number. Numbers with the same prefix are analyzed based on the length. The espace U1960 can analyze and process a regular number or prefix containing a maximum of 32 digits and an intra-office number containing a maximum of 16 digits. The espace U1960 can analyze a maximum of 1024 calling numbers and a maximum of 2048 called numbers. Prefixes can identify emergency, intra-office, local, intra-office and local, national, and international calls, and support 32 levels of customized call rights. The intra-office and local prefixes are applied when the intra-office prefix is the same as the local outgoing prefix. 14

18 4 Functions and Features A number prefix is the first digit or first few digits in a number to specify the number attribute. Prefixes can identify emergency, intra-office, local, intra-office and local, national, and international calls, and support 32 levels of customized call rights. The intra-office and local prefixes are applied when the intra-office prefix is the same as the local outgoing prefix. Voice Processing, Encoding, and Decoding The espace U1960 provides the following voice processing capabilities: Supports the voice activity detection (VAD), comfort noise generator (CNG), echo cancellation (EC), gain adjustment, jitter buffer, and packet loss compensation (PLC) technologies, providing users with high-quality voice services. Supports type of service (TOS) and Differentiated Services Code Point (DSCP) technologies, ensuring that voice streams are preferentially transmitted. Supports the Real-Time Transport Control Protocol (RTCP) and provides statistical information about the total numbers of RTP packets sent and received, total numbers of bytes sent and received, delay, jitter, and packet loss rate. The espace U1960 supports various codecs such as G.711 (A-Law/U-law), G.729a/b, ilbc, G.722, G and G In addition, it supports voice codec change and priority selection. Fax The espace U1960 supports T.30 faxes in the circuit switched domain and T.38 faxes in the packet switched domain, and transparent transmission of G.711 faxes. The end-to-end delay for transmitting signals using a fax machine cannot exceed 3 seconds. It is recommended that no more than four T.38 code switching gateways be deployed on the network. For transparent transmission, espace U19xx series IP PBX converts voice calls in G.729 encoding mode into G.711 faxes. 4.2 Supplementary Services Table 4-1 lists the supplementary services provided by the espace U1960, these features can work either in stand alone or survival mode. Table 4-1 Supplementary services Type Service Description Calling services Local Number Query Service CLIP Service CLIR Service CLIRO Service A user can dial a specified prefix to query the local number or ONLY number. When the calling line identification presentation (CLIP) service is enabled for a user, calling numbers are displayed on the user's phone. When the calling line identification restriction (CLIR) service is enabled for a user, the user's number will not be displayed on the called users' phones. If a user activates the calling line identification 15

19 4 Functions and Features Type Service Description restriction override (CLIRO) service, the user's phone can display all calling numbers (including those numbers for which the CLIR service has been enabled). CNIP Service CONP Service CNIR Service CFC Service CFU Service CFNR Service CFB Service COH Service Call Transfer Service Call Hold Service Call Park Service When a user has enabled the Calling Name Identification Presentation (CNIP) service, the name of the calling party will be displayed on the user's phone when there is a call coming. When a user has enabled the Connected Name Identification Presentation (CONP) service, the name of the called party will be displayed on the user's phone when the user makes a call. When a user has enabled the Calling/Connected Name Identification Restriction (CNIR) service, the name of this user is not displayed on the peer party's phone. If a user activates the call forwarding conditional (CFC) service, all incoming calls are forwarded to a specified number if the call meets certain conditions. The conditions include the calling number, time segment, and called user's status. If a user activates the call forwarding unconditional (CFU) service, all incoming calls are forwarded to a specified number regardless of the user status. If a user activates the call forwarding on no reply (CFNR) service, an incoming call is forwarded to a specified number if the user does not answer the call within 20 seconds. If a user activates the call forwarding busy (CFB) service, incoming calls are forwarded to a specified number when the user is busy. If a user activates the call on hold (COH) service, the third party can listen music mean while the call is on hold. If the call transfer service is enabled, a user can transfer a call to a third party by pressing the hookflash or Transfer key during a conversation. The user then automatically quits the conversation. A user can hold an ongoing call and resume it later. A user can hold a call on one phone and resume the call on another phone. If the user does not 16

20 4 Functions and Features Type Service Description Announcements Three-Party Conversation Service resume the call within the specified duration, the call is released, and the held party hears the busy tone. The espace U1960 supports phone system announcements in different languages. This version provides to the phone system music or announcements in twelve languages. The calling or called party in a two-party conversation can call a third party to start a three-party conversation or talk to the other two parties separately without ending the current conversation. Call restriction services Call back services Pickup services Call-out Restriction Service Password Change Service RCB Service CBB Service ACB Service Co-Group Pickup Service Designated Pickup Service If the call-out restriction service is enabled for a user, the user is prohibited from making specified types of outgoing calls. The password change service allows a user to change the password that is used in the call-out restriction, password-based call restriction, and ONLY services, and is used to deactivate all services. The registered call on busy (RCB) service enables a user who makes a call when the called party is busy to register the call. If the user does not perform any operations within 5 seconds the next time the user picks up the phone, a call is automatically made to the called user. If user A has enabled the Call Back on Busy (CBB) service, when user A calls user B who is busy, the system can automatically inform user A of user B's status when user B is idle. The automatic callback (ACB) service enables a user who makes a call to register the call when the called user does not answer the call. The system monitors the status of the called user. When detecting that the called user has call records, the system automatically makes a call to the calling user and establishes the connection between the called and calling users. The co-group pickup service allows a user to use the user's own phone to answer incoming calls for members in the same pickup group. The designated pickup service allows a user to use the user's own phone to answer incoming calls for another user by dialing the pickup access code and the called user's number. Secretary Secretary Service All the calls made to the manager are 17

21 4 Functions and Features Type Service Description service Secretary Station Service forwarded to the secretary, and only the secretary can put calls through to the manager. The secretary station provides the call queuing function. If a busy user has activated the secretary station service, a new incoming call can be held until the user is available. The secretary station service is used together with the secretary service. Advanced Secretary Service When a user dials a manager's phone number, the secretary's phone rings, and the secretary can decide whether to forward the call to the manager. The manager and secretary can know the peer's phone status according to the corresponding line indicator. Hotline services Delay Hotline Service If a user does not dial a number within 5 seconds after picking up a phone, a call is automatically made to connect to a preset number (hotline number) is. Attendant services Ringing services Enhanced service Instant Hotline Service Break-in Service Forced Release Service Privileged User Service Simultaneous Ringing Service Sequential Ringing Service Distinctive Ring Tone Service DND Service A user who activates the instant hotline service is immediately forwarded to a preset phone number (hotline number) after picking up the phone. An attendant can break in on the conversation between intra-office users or between intra-office and outer-office users. A three-party conversation begins. An attendant can forcibly release an ongoing conversation of a user when there is a toll call to this user or for other reasons. Users who activate the privileged user service can forcibly talk with another user. When a user who activates the simultaneous ringing service is called, the user's phone and other specified phones ring simultaneously. When a user who activates the sequential ringing service is called and does not answer the call within 20 seconds, the user's phone stops ringing and other specified phones ring in sequence. This service enables a user to determine the calling party type based on different ring tones. To block all incoming calls, a user can use the do not disturb (DND) service. If a user's phone activates the DND service, other users who call the user hear the DND announcement, and no 18

22 4 Functions and Features Type Service Description call is put through to the user. The DND service does not affect outgoing calls, and the user can call other users normally. Managemen t and setting Absent User Service Alarm Service Multi-Number Service ABD Service ONLY Service Hunt Group Service Group Call Service IP Phone Status Detection Service Remote Activation Service Service of Deactivating All Services If a user activates the user absent service, other users who call the user hear an announcement indicating that the called party is absent. This service is used when a user is unavailable for answering calls. If a user activates the alarm service, the user's phone rings automatically at the preset time. A user who activates the multi-number service can register multiple numbers including a primary number and one or more secondary numbers. The user can be reached by any of the numbers. Users can dial a one-digit or two-digit abbreviated number instead of the original called number. The One Number Link You (ONLY) service refers to a service in which multiple terminals of a user share the same number and the same supplementary services. When a user dials a hunt group number, the phones of members in the hunt group ring simultaneously or sequentially. The group call service allows the administrator to create a user group consisting of users who have enabled the instant hotline service. After an intra-group user picks up the phone, a call is automatically made to a preset outer-group user who acts as an attendant. When an outer-group user dials the group access code, all phones in the group ring. An intra-group user picks up the phone and talks to the outer-group user, and other intra-group users can pick up phones to join the call. Users can view the IP phone status of other users using the programmable key on an IP phone. A user can dial a specified prefix on a local phone to activate or deactivate forwarding services or advanced secretary services for other phones as prompted. Users can deactivate all supplementary services on their phones. The service rights are not affected 19

23 4 Functions and Features 4.3 Advanced Services The espace U1960 supports user rights management by level and call barring policies. User Rights Management by Level espace U1960 users are assigned four levels of rights from lowest to highest: default, normal, advanced, and super. Users at different levels have different rights to use supplementary services or make calls. By default, users are at the default level. Right levels for supplementary services The rights at lower levels to use supplementary services are subsets of the rights at higher levels. Table 4-2 describes the mapping between service rights and user levels. Table 4-2 Mapping between service rights and user levels Call Right User Level Default Normal Advanced Super Service Rights Users have rights to use local number query, call transfer, call forwarding, call waiting, abbreviated dialing, outgoing call barring, alarm clock, CLIP, password change, phone conference, unified access to the fax mailbox, call park, call right, ONLY, DND, and absent user services. In addition to services that users can use at the default level, users at the normal level have rights to use RCB, CBB, designated pickup, and instant conference services. In addition to services that users can use at the normal level, users at the advanced level have rights to use three-party call, call hold for multiple calls, simultaneous and sequential ringing, break-in, and forced release services. Super level: In addition to services that users can use at the advanced level, users at the super level have rights to use privileged user, secretary, and secretary station services. Right levels for call rights and call control by time segment Users at different right levels have different rights to make calls. Call rights can be control by time segment so that users at a level can make calls only in a specified time segment. Table 4-3 describes the mapping between call rights and user levels. 20

24 4 Functions and Features Table 4-3 Mapping between call rights and user levels Call Right User Level Intra-Office Call Local Call National Toll Call International Toll Call Default Yes Yes No No Normal Yes Yes Only working time No Advanced Yes Yes Yes Only working time Super Yes Yes Yes Yes Trunk preemption for higher-right-level users If trunks are insufficient, reserved trunks can be used based on the preset user level. If there is no available reserved trunk, higher-right-level users who make outgoing calls can preempt the trunks assigned for lower-right-level users. For example, a user at the super level can preempt trunks occupied by users at the advanced, normal, and default levels. When the trunk of a lower-right-level user is preempted, the user hears the busy tone (configurable), and the call is released. Users at the same level cannot preempt trunks of one another; By default, the busy tone is played when a trunk is preempted. An announcement can be also played. Call Barring Policies The espace U1960 can restrict calls as required. Table 4-4 describes the call restriction services. Table 4-4 Call restriction services Service Call barring by calling number Call barring by called number Call barring by blacklist or whitelist Description Call rights that are classified by calling number include incoming and outgoing call rights, for example, local call rights, toll call rights, and trunk call rights. Call rights that are classified by called number include incoming and outgoing call rights, for example, local call rights, toll call rights, and trunk call rights. Call barring by blacklist or whitelist allows a user to accept or reject calls made by a list of users. Users in the blacklist group can call only users in the whitelist group. Users in the ordinary call barring group can call users in the same group or in the whitelist group, but cannot call users in 21

25 4 Functions and Features Service Call barring by personal blacklist VoIP domain-based call barring Password-based call barring Call barring by card number and password Calling number authentication Call barring by region Description the blacklist group. Users in the whitelist group can call users in any of the three groups. Call barring by blacklist or whitelist allows a user to accept or reject calls made by a list of users. The personal blacklist has a lower priority than the system blacklist. IP PBX logically assigns some local users or office routes to a VoIP domain for barring calls and sets the maximum number of concurrent calls for this VoIP domain. When receiving a new call request, the IP PBX checks the calling and called parties' domains and the maximum number of calls allowed. If the maximum number of calls allowed is reached, the call fails, and the calling party hears an announcement. espace U1960 determines whether a user can make calls by asking the user to enter a password. The password is bound to the user's phone. Users can use authorized card numbers and passwords to make calls from any phones in the office. This service supports the following two dialing modes: IVR navigation dialing mode: Users enter the password-based call barring prefix, and dial a number as prompted. Full number dialing mode: Users directly enter outgoing prefix*call barring service ID*password*called number# to make a call. If the calling number authentication service is configured for the called number prefix involved in an incoming call from a local user or a trunk, the IP PBX initiates a request to the remote authentication dial-in user service (RADIUS) server to authenticate the calling number. If the authentication is successful, the IP PBX connects the call. If the authentication fails, the call fails. The IP PBX manages call rights for users who use different types of phones in various regions. After users under the IP PBX are assigned to different VoIP domains, the IP PBX determines whether a call can be 22

26 4 Functions and Features Service Anonymous call barring Restriction on outer-office call duration Description established based on the calling number, called number, or domain type. This service meets VoIP restriction requirements of India offices. Incoming and outgoing calls cannot be routed through broadband and narrowband trunks. Different rights can be assigned to trunks to prevent trunk tandem calls. This service allows the system to restrict anonymous calls (including intra-office calls and outer-office calls) to a user. When a user under espace U1960 makes an outgoing call to or receives an incoming call from an outer-office user, the IP PBX controls the call duration as configured and plays an announcement to the user when there is only one minute left. 4.4 Voice Conference Scheduled Conference espace U1960 provides scheduled and instant conferences. Users can schedule three types of conferences in the espace U1960 built-in web management system. The conference types are described as follows: Individual Dialing-In Participants dial a conference access code and enter the conference ID and password as prompted to join a conference. If the conference ID or password that a participant enters is incorrect, the participant returns to the upper-level menu. Host Convening The moderator joins a conference and performs operations on the phone to invite or remove participants. System Convening The moderator sets a participant list in advance, and the system calls the listed participants at the scheduled time. Participants can simply pick up phones to join the conference. Instant Conference Users can directly initiate an instant conference without scheduling the conference in advance. 23

27 4 Functions and Features 4.5 Voice Mailbox Services The espace U1960 provides the Voice Mailbox Services (VMS) by default. It can also connect to the Unified Messaging System (UMS) to store and manage voice messages in a unified manner to enable uses to retrieve voice messages at any time anywhere by dialing the voice message retrieving prefix. Users can leave, retrieve, delete, forward, and play voice messages and customize greetings. If a phone has a message indicator, the indicator turns on when a voice message is received. Voice mailbox services include call transfer to voice mailbox unconditional (CTVMU), call transfer to voice mailbox on no reply (CTVMNR), call transfer to voice mailbox on busy (CTVMB), call transfer to voice mailbox offline (CTVMO). The system stores a maximum of 20 voice messages for each user. The maximum duration of a voice message is 2 minutes. A maximum of 30 calls can be transferred to voice mailbox services at the same time. 4.6 Automatic Switchboard Service The automatic switchboard service is also called the interactive voice response (IVR) service. After an enterprise sets an automatic switchboard number, all incoming calls are forwarded to the automatic switchboard. The automatic switchboard can play voice prompts (customizable), collect digits, and connect calls automatically. By default, the automatic switchboard prompts calling users to enter the extension number. This feature also is support in survival mode. The automatic switchboard service provided by the espace U1960 is described as follows: A maximum of 256 automatic switchboards (ID: 0 to 255) can be configured, in which 243 automatic switchboards (ID: 12 to 254) support script customization. Each automatic switchboard can process a maximum of 240 concurrent calls depending on the media resources. Users can dial an automatic switchboard number plus an extension number to directly make a call to the extension number. This function is not supported if the automatic switchboard uses the AT0 trunk. Voice prompts can be set by time segment, that is, different voice prompts are played in different time segments. When a called party hangs up the phone on an extension, the switchboard becomes available for the calling party. 4.7 CDR The espace U1960 generates CDRs for users and write and save the CDRs in a CDR server. Other applications can obtain CDRs from a CDR server to parse them. CDR Server A CDR server stores and processes binary CDRs from multiple IP PBXs at the same time. 24

28 4 Functions and Features Operations such as querying and collecting statistics of CDRs require the support of the BMU or a third-party billing system. CDR FTP Interface CDR Console Integration FTP interfaces provided by a CDR server enable an IP PBX billing system to provide CDRs for third-party billing systems and billing centers openly and securely. A CDR server provides a graphic man-machine interface for users to manage CDRs. Querying CDRs Users can use any of the following search criteria to query CDRs from a CDR server: calling number, called number, IP address of the device for which CDRs are generated, and CDR generation time. Deleting historical CDRs Users can delete CDR files from a CDR server to free up disk space. Only CDRs that were generated one month ago can be deleted, which ensures the data security. Viewing CDR pool information Users can view the CDR pool status of an IP PBX according to the IP PBX IP address. Controlling CDR transfer Users can enable the automatic CDR transfer function. When new CDRs are generated in the CDR pool of an IP PBX, a CDR server automatically retrieves the CDRs if the automatic call transfer function is enabled. The automatic CDR transfer function is enabled by default. 4.8 Intelligent Routing The intelligent routing service automatically selects office routes when the WAN/LAN link is down to choose an IP trunk or PSTN connection trunk to provides routing polices to increase communication reliability and minimize communication costs. Routing by Time Segment Different time indexes are set for different office routes. Each time index corresponds to a specified time segment (accurate to hour). Based on the current time, the IP PBX searches for the time index for an outgoing call and selects the related office route. The IP PBX allows multiple office route selection codes to share one office route. When an office route is added to multiple office route selection codes, multiple routing policies apply to the office route. The IP PBX automatically selects the routing policy based on the outgoing prefix that a user dials. Routing by Charge Rate Different charge rate reference values are set for different office routes. The IP PBX preferentially selects the office route with the lowest charge rate reference value for outgoing calls. If all trunk circuits are busy for this office route, the IP PBX selects the office route with the second lowest charge rate reference value. Rerouting upon Call Failures 25

29 4 Functions and Features 4.9 Security When a call fails to be routed based on the office route selection code, the IP PBX selects a new route based on the standby office route selection code corresponding to the failure processing index. Route Load Balancing The IP PBX balances traffic among multiple preset routes. The IP PBX polls office routes based on office route IDs in ascending order till an office route that contains idle circuits is found. The later calls will poll from the next office route after an office route is selected by the preceding call. Routing by Percentage Different percentages are set for different office routes. The IP PBX selects office routes in turn based on the preset percentages. Except the office routes whose percentages are set to 100%, office routes are polled based on preset percentages in descending order. Routing by User Right Level Different office routes are set for different user right levels. When a higher-right-level user makes an outgoing call, the IP PBX preferentially selects the office route that is set for the user right level. If selecting the office route fails, the IP PBX selects the office route that is set for a lower user right level. When routing by user right level is used, the IP PBX preferentially uses an office route at the user right level to route calls. If no circuit is idle, the office route that is set for a lower user right level is used. Office routes that are set for the same level are polled based on office route IDs in ascending order. To ensure good voice quality of calls for higher-right-level users, these users are allocated with high-performance office routes. Lower-right-level users, however, are not allowed to use these office routes. Trunk Link Balancing The IP PBX preferentially selects a trunk with more idle circuits to balance loads among available trunk links. Local and Survival Mode The local and survival mode connection and routing policies for IP trunk or PSTN trunk are supported in the espace U1960 Gateway Anti-attack Measures Web Security Measures The espace U1960 provides enhanced security measures for users. The communication port matrix is delivered with the product documentation. Only services and ports listed in the communication port matrix need to be enabled. The communication port matrix contains the following information: available ports, transport layer protocols used by the ports, network elements (NEs) that use the ports to communicate with peer NEs, application layer protocols used by the ports and description of the services at the application layer, information about whether services at the application layer can be disabled, authentication modes of the ports, and functions of the ports (such as transmitting management traffic, control traffic, or data traffic). The Secure Shell (SSH) protocol is used to authenticate user login. Media streams and TLS signaling in the voice services are encrypted to ensure user data security, by using the AES and DES standards. This feature is support in local and survival mode. The system has the following password rules: Prompts users to change passwords at first login. 26

30 4 Functions and Features Prevents violent password cracking, and locks accounts or IP addresses after a specified number of verification code or password attempts. Transmits user names and passwords using the HTTPS. The system provides the following authentication functions: Verifies the user session ID and user rights for each request to access a page or servlet that requires authorization. Executes the final authentication on users on the server. Verifies data generated by users on the server, and encodes data using the HTML before transmitting the data to clients, which prevents malicious code and cross-site scripting attacks. Uses the web security scanning software to scan web servers and applications, which prevents high-level vulnerabilities. 27

31 5 Operation, Maintenance, and Management 5 Operation, Maintenance, and Management About This Chapter The espace U1960 can be managed using the web management system and espace LMT (LMT for short). Operations such as conference management and service registration/deregistration can be performed using the web self-service system. Also, the espace U1960 can be operated and maintained using commands. 5.1 Web Management System The web management system is a built-in configuration and monitoring tool of the espace U1960. Users can access the system for configuration management and resource query. 5.2 Web Self-Service System The espace U1960 provides a simple web self-service system. Common users can access the system using a browser to perform operations such as managing conferences and registering or deregistering services. 5.3 LMT The Local Maintenance Terminal (LMT) is a management system for the espace U1960. It provides various functions such as alarm management, configuration management, signaling tracing, upgrade, log collection, and offline operations. 5.4 CLI Management Mode Users can use the command line interface (CLI) to operate and maintain the espace U Web Management System The web management system is a built-in configuration and monitoring tool of the espace U1960. Users can access the system for configuration management and resource query. Table 5-1 describes the functions provided by the web management system. 28

32 5 Operation, Maintenance, and Management Table 5-1 Functions provided by the web management system Management Function Subfunction Configuration management Provides a configuration wizard for user and trunk configurations. Provides graphical user interfaces (GUIs) for data, service, and network configurations. Resource query Monitors board status. Monitors CPU usage. Monitors E1/T1 status. Monitors channel status. Monitors trunk status. 5.2 Web Self-Service System The espace U1960 provides a simple web self-service system. Common users can access the system using a browser to perform operations such as managing conferences and registering or deregistering services. Table 5-2 lists services provided by the web self-service system. Table 5-2 Services provided by the web self-service system Service Scheduling conferences Viewing conferences Registering services Description Users can schedule conferences as required and add participants. Users can view and join conferences. Users can register or deregister the ONLY, call forwarding, and voice mailbox services. The web self-service system supports only Internet Explorer 6 or later, and supports the 1024 x 768 resolution or higher. The recommended resolution is 1280 x The Internet Explorer version must support ActiveX and Javascript. 5.3 LMT The Local Maintenance Terminal (LMT) is a management system for the espace U1960 and the IP-PBX U1980. The e-sight platform can managed local or remote, by schedule or automatic, this includes DID s, Dialing Plan, etc. 29

33 5 Operation, Maintenance, and Management The LMT connects to the espace U1960 using SSH/Telnet. A maximum of four espace U1960s can be connected to the LMT. Table 5-3 describes the management functions provided by the LMT. Table 5-3 Management functions provided by the LMT Management Function Subfunction Alarm management Displays and queries real-time alarms. Displays and queries historical alarms. Queries alarms by parameters such as the alarm severity, alarm type, alarm ID, time, and device IP address. Collects statistics on alarm by level. Acknowledges and clears alarms. Synchronizes and filters alarms. Exports alarms. Configuration management Adds, deletes, modifies, and logs in to the espace U1960. Configures data and queries status through the command navigation tree. Manages voice files. Manages patches and versions. Manages licenses. Provides a built-in TFTP server. Generates the VU script on the GUI. Integrates with the MakeTone tool. Signaling tracing Traces and parses SIP broadband signaling. Traces and parses PRI, SS7, R2 and QSIG narrowband signaling. Reports the traced messages with a user phone number being the tracing condition. Manages tracing tasks. Displays messages and signaling in real time. Automatically saves messages and signaling. Queries historical messages and signaling. Deletes messages or signaling. Upgrade Uploads espace U1960 version files. Backs up version files and patches. Upgrades the espace U1960 and the data.bin file. Rolls back the espace U1960. Generates upgrade logs. Displays the upgrade progress dynamically. Log collection Downloads operation logs. Downloads run logs. 30

34 5 Operation, Maintenance, and Management Management Function Subfunction Queries and retrieves debug logs. Offline operation Parses alarm files, log files, and signaling tracing files offline. Upgrades data files offline. 5.4 CLI Management Mode Users can use the command line interface (CLI) to operate and maintain the espace U1960. Figure 5-1 shows the network of the CLI management mode. Figure 5-1 Network Users can use the CLI in either of the following ways: Connect a PC directly to the serial port of the espace U1960, use the Super Terminal tool to set up a connection between the PC and the espace U1960, and then configure or commission the system. Use SSH to log in to the espace U1960 through the service network port from a PC, and then configure or commission the system. 31

35 6 Technical Specifications 6 Technical Specifications About This Chapter This topic provides the key technical specifications of the espace U Physical Specifications Performance and Capacity Ports and Protocols Standard Compliance 6.1 Physical Specifications Table 6-1 lists the dimensions, weight, power supply settings, power consumption, and operating environment of the espace U1960. Table 6-1 Physical specifications Item Dimensions (H x W x D) Weight Maximum power consumption in full configuration Input voltage (AC power) Power frequency (AC power) Maximum output power (AC power) Input voltage (DC power) Input electricity (DC power) Maximum output power (DC power) Indicator 86.1 mm x 442 mm x 310 mm 10 kg (weight in full configuration) 220 W V AC 50 Hz/60 Hz 300 W V DC to -60 V DC; typical value: -48 V DC 9 A 350 W 32

36 6 Technical Specifications Item Storage temperature Long-term operating temperature Short-term operating temperature Relative humidity Ambient particulate concentration Indicator -40 C to +70 C 0 C to 45 C -5 C to +55 C NOTE Short-term running means that the device works nonstop for not greater than 48 hours and the cumulated working time of the device in a year does not exceed 15 days. 5% to 95% (non-condensing) < 180 mg/m Performance and Capacity Table 6-2 lists the performance and capacity for the espace U1960. Table 6-2 Performance and capacity Parameter Setting Max. users 1000 Max. Concurrent call compression channels 300 Max. capacity of AT0 trunks 60 Max. capacity of PRI trunks (E1) Max. capacity of PRI trunks (T1) Max. capacity of QSIG trunks (E1) Max. capacity of QSIG trunks (T1) Max. capacity of SS7 trunks (E1) Max. capacity of R2 trunks (E1) Max. capacity of SIP trunks 240 Busy Hour Call Completion 36K 33

37 6 Technical Specifications Parameter Max. number of concurrent incoming/outgoing calls through digital trunks Max. number of concurrent incoming/outgoing calls through SIP trunks Max. number of voice messages a user can have Max. time length of a voice message Max. capacity of a CF card Maximum number of CF cards Setup time of a fax call Max. pages of standard fax sample paper Conference processing capability Max. number of concurrent SRTP calls Max. number of SIP trunks encrypted by TLS Setting s 8 GB 3 Total storage space of a voice mailbox = [Data size of 1-second voice (8 KB) x Max. voice message time length (120s) + File header size (0.1 KB)] x Max. voice message number (20) x User number. A CF card can provide storage services for about 430 voice mailboxes. < 20s 20 Supports a maximum of 120 conferences. Supports a maximum of 360 participants. A single conference supports a maximum of 60 participants. Each MTU board supports a maximum of 120 calls Ports and Protocols Ports The espace U1960 series products provide FE ports, E1/T1 digital trunk ports, and debugging ports. Table 6-3 lists the external ports of the espace U1960 series products. 34

38 6 Technical Specifications Table 6-3 External ports Port Board Quanti ty Function GE port SCU 3 Connects the espace U1960 to a LAN and functions as an IP service port. The espace U1960 can work in single-network-port, dual-network-port mode, or triple-network-port mode, under TCP/IP protocol. USB port SCU 2 Used for future expansion. E1/T1 port MTU 2 Connects the espace U1960 to digital trunks of an upper-level office, for example, LE. FXS port ASI 32 Connects to a maximum of 32 analog phones. OSU 12 Connects to a maximum of 12 analog phones. FXO port OSU 12 Connects to a maximum of 12 analog trunks. Debugging port SCU 1 Configures and commissions the espace U1960. MTU 1 The debugging port is an RS-232 serial port (RJ45 jack). ASI 1 OSU 1 Signaling and Protocols Figure 6-1 shows the distribution of main signaling and protocols. 35

39 6 Te chnical Specifications Figure 6-1 Distribution of signaling and protocols Table 6-4 lists the signaling and protocols supported by the espace U1960. Table 6-4 Signaling and protocols Signaling/Protoco l SS7 PRI R2 Function Enables the communication between the espace U1960 and switches supporting the SS7 signaling and enables the espace U1960 to access E1 trunks provided by the switches. Enables the communication between the espace U1960 and the switches on the ISDN and enables the espace U1960 to access E1/T1 trunks provided by the switches on the ISDN. Enables the communication between the espace U1960 and traditional switching devices and enables the espace U1960 to access E1 trunks provided by the traditional switching devices. QSIG Enables the communication between the and switches supporting the QSIG signaling and enables the to access E1 trunks provided by the switches. SIP Enables the interconnections between espace U1960s and connects the espace U1960 to IADs and SIP multimedia packet terminals. Telnet Connects LMTs or remote operation and maintenance terminals to espace U1960s. 36

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