PlexTel VoIP-PBX system A4 -2006_Edit

Full-featured VoIP system

∗Flex-1 for HOME/SOHO

∗Flex-2 for small-medium office

∗Precise-1 for enterprise

∗Precise-2 for large enterprise

Year 2006 Product line

Poise Technology

IP-PBX System

Enterprise class IP-PBX system

IP-PBX System

Overview All Plextel IP-PBX series is a feature-rich, highly-scaleable VoIP PBX based. It provides you with a business-class PBX system with advanced features such as least-call-cost-route, internet calling, auto-attendant, voice mail with email mes-sage notification, automatic call distribution (ACD), call re-cording, call conferencing and peer IP-PBX support, that are only available to high-end PBX but at a lower cost.

Easy Access and Control Plextel provides an intuitive web-based interface for the administrator to configure and manage the PBX. The end-user may also access their voice messages or change their user options using a web-browser.

IP Networking One of the key benefits of IP Networking is reduced call costs. Internal voice calls can be carried

over your private local or wide-area IP network (LAN or WAN) or shared internet network. National and inter-national call charges can be reduced dramatically using compression technology which, compared to tradi-tional voice calls, allows greater number of VoIP calls to be carried over the same facilities.

To Order Contact: [email protected]

Benefit

• offer advance features such as such as auto attendant, ACD, call recording and call confer-encing that are only available to high-end PBX but at a lower cost.

• based on standard-based technologies (i.e. H323, SIP, G.711, etc), This means no more lock-in to a single Phone vendor.

• support VoIP Nat (Network Address Translation) friendly protocol, Much less trouble using the shared internet system

• free intra-network calling among geographically dispersed offices (eliminate long-distance charge)

• single network infrastructure for both voice and data communication, thus resulting in cost savings in installation, configuration, and management

• flexible/programmable dial-plan mean no lost contact with your customer • enable mobile workforce and remote staff to connect to the PBX seamlessly as if they are in

the office • easier to expand (no hardware upgrade required for adding extensions) • optional voice interactive system for add-on interactive voice applications

Call Features Alarm Receiver Authentication Automated Attendant Blacklists Blind Transfer Call Detail Records Call Forward on Busy Call Forward on No Answer Call Forward Variable Call Monitoring Call Parking Call Queuing Call Recording Call Retrieval Call Routing (DID & ANI) Call Snooping Call Transfer Call Waiting Caller ID Caller ID Blocking Caller ID on Call Waiting Calling Cards Conference Bridging Database Store / Retrieve Database Integration Dial by Name Direct Inward System Access Distinctive Ring Do Not Disturb

Flexible Extension Logic Interactive Directory Listing Interactive Voice Response (IVR) Local and Remote Call Agents Music On Hold Music On Transfer Random or Linear Play Volume Control Overhead Paging Protocol Conversion Remote Call Pickup Remote Office Support Roaming Extensions Route by Caller ID Supervised Transfer Three-way Calling Time and Date Transcoding Trunking VoIP Gateways Voicemail Visual Indicator for Message Waiting Stutter Dialtone for Message Waiting Voicemail to email Voicemail Groups Web Voicemail Interface

Telephony Interoperability

E&M E&M Wink FXS FXO E1

Protocols

H323 SIP (Session Initiation Protocol) IAX (Inter-Asterisk Exchange)

Codec

G.711 (A-Law & u-Law) G723.1 (pass through only) G.726 G.729 (Optional) GSM iLBC LPC-10 Speex

Interface Available

Quad-Port FXS, FXO Quad-Span togglable E1/T1/J1 Dual-Span togglable E1/T1/J1 Single-Span togglable E1/T1/J1



Flex-1 The Flex-1 is an affordable system which contains highly functional IP-PBX Voice Server. With its small form factor and low power consumption, it will be best suite for Home, Small office or SOHO in which the Intelligence Voice technology is important. Same as every Flex family, Flex-1 Voice Server combines the functionality of a PBX, SIP proxy, Voice Mail server, and more. Key features include Graphic User Interface (GUI), call groups, auto-attendant, paging, music-on-hold, and conference bridging.

*Flex-1 is designed to handle for up to 20 Concurrent Calls.

* Flex-2 is designed to handle for up to 40 Concurrent Calls.

* Precise-1 is designed to handle for up to 80 Concurrent Calls.

Flex-2 The Flex-2 is a high performance system which contains highly functional IP-PBX Voice Server. With its highly expandability and low power consumption, it will be excellent for advance SOHO, SME in which the Intelligence Voice technology is important. Flex-2 Voice Server combines the functionality of a PBX, SIP proxy, Voice Mail server, and more. Key features include Graphic User Interface (GUI), call groups, auto-attendant, paging, music-on-hold, and conference bridging.

Precise-1 The Precise-1 is an Enterprise-level system which contains highly functional, and highly customizable IP-PBX Voice Server. Precise-1 was designed with the rugged dependability of a specially configured industrial strength hardware platform. It is a great foundation for enterprise that are ready to make the move to VoIP, Internet telephony technology. Precise-1 Server combines the functionality of a Router, PBX, SIP proxy, Voice Mail server, and more.

* Precise-2 is designed to handle for up to 150 Concurrent Calls.

Precise-2 The Precise-2 is an Enterprise-level system which contains highly redundant, highly functional, and highly customizable IP-PBX Voice Server. Precise-2 was designed with the rugged dependability of a specially configured industrial strength hardware platform. It is a great foundation for enterprise that are ready to make the move to VoIP, Internet telephony technology. Precise-2 Server combines the functionality of a Router, PBX, SIP proxy, Voice Mail server, and more.

Optional Interface:


TE405P Quad-Span togglable E1/T1/J1 card enables per card or per-port selection of either T1, E1, and J1 signaling formats (5.0 volt PCI only).

TE410P Quad-Span togglable E1/T1/J1 card enables per card or per port selection of either T1, E1, J1 signaling formats (3.3 volt PCI only).

TE411P Quad-Span togglable E1/T1/J1 echo cancellation card enables per card or per port selection of either T1, E1, J1 signaling formats (3.3 volt PCI only).

TE406P Quad-Span togglable E1/T1/J1 echo cancellation card enables per card or per port selection of either T1, E1, J1 signaling formats (5.0 volt PCI only).

* T1 support 24 channels * E1 support 30 channels





TDM400P Quad-Port half-length PCI card which supports standard analog or ADSI telephones and regular POTS lines.

TE110P Single-Span selectable T1/E1/J1 half-length (available with 2U bracket) PCI card sporting the same features as the TE410P.

TE210P / TE205P Dual-Span togglable E1/T1/J1 card enables per card or per port selection of either T1, E1, J1 signaling formats (3.3/ 5 volt PCI only).

TDM2400P Full-length 32-bit 33MHz PCI 2.2-compliant card that supports quad FXS station and quad FXO office interfaces (for a total of 24 maximum channels on a single PCI slot) for connecting analog telephones.



* Support upto 24 analog lines

* Support upto 4 analog lines

Phone / Phone Adapter:


IP Phone 360 Designed for the maximum productivity and efficiency in the every business environment. Dedicated keys for audio and call control, Context-sensitive menus for additional Functionality at the moment you may need. 47 keys, 13 LEDs, 12 Programmable function keys. Headset Connection, Dual Ethernet Connection. PPP0E, SIPS RFC3261, STUN, ENUM, NAT, ICE. Compression G.723.1 and other. Security: SIPS/SRTP.

* Additional keypads with 42 programmable function keys. * Tiltable graphic display (128X64 pixels). * POE Support

IP Phone 320 Idea for general office and knowledge-worker environments, the 320 is an affordable and powerful SIP business telephone with built-in, full-duplex speakerphone and three-party conference bridging. 47 keys, 13 LEDs, 12 Programmable function keys. Phone, Headset Connection, Dual Ethernet Connection. PPP0E, SIPS RFC3261, STUN, ENUM, NAT, ICE. Compression G.723.1 and other. Security: SIPS/SRTP.

* Two-Line display with graphical field. * POE Support

IP Phone 300 Basic model of business telephone family, 300 fulfils the most important requirements of VoIP telephony and additionally offers numerous functions that are indispensable in the business world. 27 keys, 6 programmable function keys, Headset Connection, Dual Ethernet Connection, SIPS RFC3261, STUN, ENUM, NAT, ICE. Compression G.723.1 and other. Security: SIPS/SRTP, TLS.

* Two-Line display with graphical field.

IP Phone AT-320PD The internet based voice network phone terminal supporting power supply through Ethernet, adopts multiple voice control protocols and voice compressing coding methods to directly convert analog voice into packet for internet transport, thus effectively using the existing bandwidth to provide PSTN quality voice service. Support H.323 V4, MGCP, SIP, IAX2, Net2phone private protocol. Audio codec G.711, G.723, G.729.

* 2 X RJ45 Connector * Power Over Ethernet (POE) Support


* 2 X RJ45 Connector

IP Phone AT-320ED The internet based voice network phone terminal, adopts multiple voice control protocols and voice compressing coding methods to directly convert analog voice into packet for internet transport, thus effectively using the existing bandwidth to provide PSTN quality voice service Support H.323 V4, MGCP, SIP, IAX2, Net2phone private protocol. Audio codec G.711, G.723, G.729

IP Phone 480i An advanced, fully featured, IP Screen Telephone that provides a flexible IP solution designed with interoperability and ease of use in mind. Support IETF SIP (RFC3261), IETF MGCP (RFC3435), G.711 μ–law/A-law, G.729

IP Phone 480i CT The 480i-CT offers cordless mobility for enterprise workers. The cordless handset provides VoIP access anywhere within an office or warehouse with exceptional features and performance. Support IETF SIP (RFC3261), IETF MGCP (RFC3435), G.711 μ–law/A-law, G.729

* 1 X RJ45 Connector

* Cordless Mobility supports up to 4 cordless headsets. * Large Base Unit Display. * Multiple Line/Call Appearance Lights. * Shared Call Appearances. * Excellent voice clarity and delivery * Built in dual 10/100 switched Ethernet ports and support POE.

IP Phone AT-320EE The internet based voice network phone terminal, adopts multiple voice control protocols and voice compressing coding methods to directly convert analog voice into packet for internet transport, thus effectively using the existing bandwidth to provide PSTN quality voice service Support H.323 V4, MGCP, SIP, IAX2, Net2phone private protocol. Audio codec G.711, G.723, G.729

* Eight-line backlit screen with six context sensitive softkeys. * Large storage for personal directory callers logs, and redial list. * Support up to 9 simultaneous calls. * Shared calls allow you to place a call on hold at one set and pick it up easily at another set. * Firmware upgradeable. * Built in dual 10/100 switched Ethernet ports and support POE.


IP Phone 9133i An advanced, fully featured multi-line IP Telephone that takes full advantage of VoIP technology by offering a flexible, interoperable solution at an affordable. Support IETF SIP (RFC3261), G.711 μ–law/A-law, G.729.

* Large storage for personal directory, callers log, and redial list. * Support up to 3 simultaneous calls. * Quality speakerphone with excellent voice clarity and delivery. * Excellent voice clarity and delivery * Firmware upgradeable. * Built in dual 10/100 switched Ethernet ports and support POE.

IP Phone 9112i A value priced, basic single line IP Telephone with speakerphone offering a flexible, interoperable IP solution that meets VoIP budgets for most applications. Support IETF SIP (RFC3261), G.711 μ–law/A-law, G.729.

* Large storage for personal directory, callers log, and redial list. * Excellent voice clarity and delivery * Firmware upgradeable.

IP Phone 360 Expansion Module With the IP-Phone 360 Expansion Module, your 360 has the best features to meet the requirements of executive and administrative staff who require high call coverage and flexibility from communication systems. Using your 360 with the Expansion Module, call coverage and management is even more easy and efficient to handle. With the expansion module, you add 42 additional programmable keys and LEDs, increasing the total number of keys and LEDs to 54. More comfort is provided by the built-in web interface of your 360 VoIP phone. With the built-in web interface, you can easily configure each function key according to your preferences.on-hold, connected call etc., will be shown by the LEDs associated with each key.

Headsets HS-MM2 HS-MM2 Headset have been updated with a brand new look and improved ergonomics and usability. With the headsets, we provides a complete product line for use in a variety of Internet telephony environments: SMEs, home offices, and call centers.

* Flexible metal mic boom. * Quick-disconnect cord. * RJ-11 modular plug. * Adjustable headband. * Noise-canceling microphone boo, * Superb audio quality. * Light weight (95g).


ATA Adapter AG-168V The internet based one port voice gateway, adapts multi voice control protocols and voice compression codec to directly convert analog voice compression coded to directly convert analog voice into ip packet for internet transport, thus effectively using the existing bandwidth to provided PSTN quality voice service. Support H323, SIP, MGCP, IAX2 and other protocols.

* 1 X RJ45 Connector + 2 X RJ11 Connectors * Able to dial both IP phone and PSTN.

ATA Adapter AG-468V The multi-function voice/fax/data access gateway specially designed for operators, enterprises and household users Supports standard SIP protocol, multiple network access (ADSL, Cable Modem, etc.), suitable for operators and virtual operators to offer centralized number sale, IP telephone supermarket and enterprise virtual telephone network service. Supports SNMP protocol

ATA Adapter AG-268 The multi-function voice/fax/data access gateway specially designed for operators, enterprises and household users. Supports standard SIP protocol, multiple network access (ADSL, Cable Modem, etc.), suitable for operators and virtual operators to offer centralized number sale, IP telephone supermarket and enterprise virtual telephone network service. Supports SNMP protocol

* 2 X RJ45 Connector + 2X RJ11 Connector * Build-in routing/NAT function * Able to provide dual-server registration

* 1 X RJ45 Connector + 4 X RJ11 Connector * Supports various private network transparency modes. * Able to provide dual-server registration

USB Gateway AU-600 The latest USB interface voice gateway. It supports call transfer between Skype and PSTN and secondary dialing. AU-600 enables ordinary desktop telephone to dial and receive Skype calls. When the user is not available, AU600 can forward SKYPE call to mobile phone or any other specified number. With AU600 installed on one’s computer (at stand-by mode), the user can use mobile phone to dial pre-set user on Skype.

* 1 X USB 1.1 + 2 X RJ11. * Support Skype


USB Phone AU-200 The USB interface portable phone. It offers USB1.1 interface and built-in 16bit sound card, and is equipped with clear-bimap LCD to support four line display and multiple languages. AU-200 offers humanistic key operation, and functions such as hand-free, mute, redial/dial-back, caller ID, caller statistics, in-call volume adjustment and personal ringing tone.

USB Phone AU-100 The USB2.0 interface portable phone and built-in 16bit sound card, and can be installed without driver under win98SE/win2000/win xp /Mac OS9/Mac OS X. AU100 offers open SDK to simplify software development and specialized communication applications based on it.

* Support Skype. * Excellent voice quality. * Rich feature.

* It’s can support various popular IM software such as Skype, SJ phone, X-Lite, Yahoo, VoipBuster... It supports E.164 based dialing rule, has simple and easy-to-use key operation. It is a very cost-effective voice product.

Web: www.poisetechnology.com Email: [email protected]

76 Sukhumvit Soi 2 Klongtoey, Bangkok Thailand 10110 Tel: (662)-656-8598 Fax: (662)-250-9769

Poise Technology

PlexTel VoIP-PBX system A4 -2006_Edit

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