VOIP GATEWAY HG-325 for Home & Small Office by OpticLAN

OpticLAN has developed and launched a newest product: HG325 to meet SOHO and home user′s access requirement, It provides VoIP and LAN interconnect services in one box. HG325 softly support SIP protocol, integrates rich interface (2*FXS+1*WAN+4*LAN).

Technical Specs:

1）CPU: MIPS SOC GC3210W, Maximum Clock frequency 266MHz;

2）Memory: 4MBytes NOR Flash, 32MBytes SDRAM;

3）Operating System: Linux2.6;

4）POTS: 2FXS, RJ11, can connect traditional telephone, Fax machine or MODEM;

5）Ethernet interface: 5RJ45(1 WAN, 4 LAN), complying with IEEE802.3/802.3u, 10/100M auto-adaptive, Full Duplex, support Auto MDIX, VLAN Tag, IEEE802.3x flow control and so on, could connect PC, Switch, Router, Firewall and so on;

6）Indicator: total 8, 1 for power supply, 2 for configuration status, 2 for telephone status and 5 for Ethernet status;

7）Button: 1, for reset or restore to factory configuration;

8）Power supply: direct current +12V, supply from external power adapter;

9）Power consumption: 6W with full run;

10）Ambient temperature: -5 deg C ~ +55 deg C;

11）Ambient humidity: 10~90 ％ , non-condensation.

Features:

1）Protocol: SIP (RFC2543, RFC3261), softly compatible with widely used Soft-Switch. such as Huawei, ZTE, Alcatel-Lucent;

2）Voice Coder: ITU-T G.711(A/ m -law), G.723.1(5.3K/6.3K), G.729A/B, HG325 can auto negotiate with SIP Server;

3） Echo canceller: complied with ITU-T G.168-2002, 128ms(Max);

4）Network protocol: TCP/IP, UDP, RTP/RTCP, ARP, TFTP, HTTP, Telnet, ICMP, IGMP, NTP, SNTP and so on;

5）Powerful data switch function: VLAN, IGMP, 4 levels QOS, port rate control and so on,

6）IP address distribution: static or dynamic(PPPOE and DHCP);

7）NMS: support CLI and Web, integrates IVR, user can set HG325 by the connected phone dial pad;

8）Equipment can be remotely upgraded through WAN(Internet);

9）Volume gain: -31dB ~ +31dB;

10）Support silence compression, voice active detection(VAD) and comfortable noise generation (CNG);

11）Support T.30, T.38 fax, fax Pass-Through mode;

12）Support Modem dial;

13）Support auto-adaptive Jitter-Buffer;

14）Support TOS(Type Of Service), QOS(Quality of Service);

15）Support RFC2833 and redundant RFC2833;

16）Support PLC(Packet loss compensation).

VOXDE UC-800 IPPBX

VOXDE   UC-800 (Made in Taiwan)  is   an   IP   PBX   equipment   designed   for   SOHO,Small   and   Medium   Business   applications.  VOXDE UC-800      integrates up to 8 analog ports [ 6 CO Trunk / 2 Analog port or 4 CO Trunk / 4 Analog port) to support the existing analog phone lines or PSTN trunks. It provides Ethernet port for the connection to the Internet   Telephone   Service   Providers   [ITSP]   to   cut  telephone bill   when   long   distance   calls   are   routed through the Internet. VOXDE UC-800 provides the full PBX features including auto attendant, conferencing, voice   mail,   IVR,   call   forward   on   busy   or   no   answer,   blind   transfer,   supervised,  transfer,   call parking   / queuing,   music   on   hold,   remote   office   support,   etc.   It   has   a   built-in   SIP   proxy   and  supports   extension registrations and configurable dial plan. It supports SIP compliant devices as peers: IP phones, soft phones (xlite),  ATA,  Nokia  cell  phone,  IPhone,  Android   and  voice  gateways.   Web   based   configuration  and management are provided in the VOXDE UC-800. Web call feature is provided in the VOXDE UC-800.

 HARDWARE SPECIFICATIONS  
- 1 Ethernet port 
- 2 FXS / 6 FXO ports on base unit 
- 4 Configurable FXS/FXO ports 
- LED: Power, Alarm and 8 LED for FXS/FXO ports 
- Mechanical Dimension (L*W*H):165*143*25  mm 
- Power: + 12V DC 
- Temperature : 0 to 45 
- Humidity : up to 90 % non-condensing  

Features

    - Build-in SIP proxy server, support SIP compliant devices as peers: IP phones, soft phones, ATA and             voice gateways
    - Supports ITU G.711, G.726, G729, G723.1, and GSM voice codecs and echo cancellation
    - Supports basic calls, call transfer, forwarding, parking, pickup, and hunting
    - Supports voice mails, conference room
    - Supports Remote Office connections
    - Supports system expansions (Inter PBX)
    - Configurable dial plan
    - Make phone calls via SIP service providers
    - Web based configuration and management
    - Provide Web Call service (option)
    - Support recording to FTP Server.

Linksys SPA942 4-Line IP Telephone with 2 Port Ethernet Switch, PoE and Hi-Res Display

Linksys SPA942

4-Line IP Telephone with 2 Port Ethernet Switch, PoE and Hi-Res Display

4-Line IP Telephone with 2 Port Ethernet Switch, PoE and Hi-Res Display Advanced, Affordable, Feature Rich IP Phone for the Home Office and Business

Stylish and functional in design, the SPA942 VoIP telephone is ideal for a residence or business using a hosted IP telephony service, an IP PBX, or a large scale IP Centrex deployment. The SPA942 leverages industry leading VoIP technology from Linksys to deliver an upgradeable high quality IP Phone that is unparalleled in features, value, and support. Standard features on the SPA942 include four active lines, dual switched Ethernet ports, 802.3af PoE support, a high resolution graphical display, high quality, full-duplex speakerphone, and a 2.5 mm head-set port. Each line can be independently configured to use a unique phone number (or extension), or can be configured to use a shared number that is assigned to multiple phones.

Comprehensive Interoperability and SIP Based Feature Set Based on the SIP standard, the SPA942 has been tested to ensure comprehensive interoperability with equipment from VoIP infrastructure leaders enabling service providers to quickly roll-out competitive, feature rich services to their customers. With hundreds of features and configurable service parameters, the SPA942 addresses the requirements of traditional business users while leveraging the advantages of IP telephony. Features such as easy station moves, presence, and shared line appearances (across local and geographically dispersed locations) are just some of the many advantages of the SPA942.

Carrier-Grade Security, Provisioning, and Management

The SPA942 uses standard encryption protocols to provide secure remote provisioning and unobtrusive in-service software upgrades. Linksys secure remote provisioning tools include detailed performance measurement and troubleshooting features, enabling network providers to deliver high quality support to their subscribers. Remote provisioning also saves service providers the hassle and expense of managing, pre-loading, and re-configuring customer premise equipment (CPE).

Benefits:

* Full featured four-line business class IP Phone supporting Power over Ethernet 802.3af

* Connect directly to an Internet Telephone Service Provider or connect to an IP PBX

* Dual switched Ethernet ports, Speakerphone, Caller ID, Call Hold, Conferencing, and more

* Easy installation and secure remote provisioning. Menu based and web based configuration.

* An optional power supply is orderable separately for this product if not utilizing Power over Ethernet(PoE)

Features:

* Up to Four Call Appearances with Independent Configuration and Registration

* The SPA942 ships with two line appearances enabled. A two line upgrade is available via a software license key installed locally using the SPA942 web interface, or installed remotely via a secure profile update.

* Pixel Based Display: 128x64 Monochrome Graphical Liquid Crystal Display (LCD)

* Line Status - Active Line Indication, Name and Number

* Menu Driven User Interface

* Digits Dialed with Number Auto-Completion

* Shared Line Appearance **

* Speakerphone

* Call Hold

* Music on Hold **

* Call Waiting

* Caller ID Name and Number and Outbound Caller ID Blocking

* Outbound Caller ID Blocking

* Call Transfer - Attended and Blind

* Call Conferencing

* Automatic Redial

* On-Hook Dialing

* Call Pick Up - Selective and Group **

* Call Park and UnPark **

* Call Swap

* Call Back on Busy

* Call Blocking - Anonymous and Selective

* Call Forwarding - Unconditional, No Answer, On Busy

* Hot Line and Warm Line Automatic Calling

* Call Logs (60 entries each): Made, Answered, and Missed Calls

* Redial from Call Logs

* Personal Directory with Auto-dial (100 entries)

* Do Not Disturb (callers hear line busy tone)

* URI (IP) Dialing Support (Vanity Numbers)

* On Hook Default Audio Configuration (Speakerphone and Headset)

* Multiple Ring Tones with Selectable Ring Tone per Line

* Called Number with Directory Name Matching

* Call Number using Name - Directory Matching or via Caller ID

* Subsequent Incoming Calls with Calling Name and Number

* Date and Time with Intelligent Daylight Savings Support

* Call Duration and Start Time Stored in Call Logs

* Call Timer

* Name and Identity (Text) Displayed at Start Up

* Distinctive Ringing Based on Calling and Called Number

* Ten User Downloadable Ring Tones - Ring Tone Generator Free from www.linksys.com

* Speed Dialing

* Configurable Dial/Numbering Plan Support-per line

* Intercom **

* Group Paging **

* DNS SRV and Multiple A Records for Proxy Lookup and Proxy Redundancy

* Syslog, Debug, Report Generation, and Event Logging

* Secure Call Encrypted Voice Communication Support

* Built-in Web Server for Administration and Configuration with Multiple Security Levels

* Automated Remote Provisioning, Multiple Methods. Up to 256 Bit Encryption: (HTTP, HTTPS, TFTP)

* Optionally Require Admin Password to Reset Unit to Factory Defaults

** Feature requires support by call server

1. Handset

2. LCD screen (SPA962 in color)

3. Voice mail indicator light

4. Lines (SPA941, SPA942 and SPA962)

5. Soft buttons

6. Navigation button 7. Mailbox button

8. Hold button

9. Setup button

10. Volume adjustment bar 11. Headset button

12. Speaker button

13. Mute button

14. Dial pad

4-Line IP Telephone with 2 Port Ethernet Switch, PoE and Hi-Res Display
Specifications
Model	SPA942
Data Networking	• MAC Address (IEEE 802.3) • IPv4 - Internet Protocol v4 (RFC 791) upgradeable to v6 (RFC 1883) • ARP - Address Resolution Protocol • DNS - A Record (RFC 1706), SRV Record (RFC 2782) • DHCP Client - Dynamic Host Configuration Protocol (RFC 2131) • ICMP - Internet Control Message Protocol (RFC792) • TCP - Transmission Control Protocol (RFC793) • UDP - User Datagram Protocol (RFC768) • RTP - Real Time Protocol (RFC 1889) (RFC 1890) • RTCP - Real Time Control Protocol (RFC 1889) • DiffServ (RFC 2475), Type of Service - TOS (RFC 791/1349) • VLAN Tagging 802.1p/q - Layer 2 QoS • SNTP - Simple Network Time Protocol (RFC 2030)
Voice Gateway	• SIPv2 - Session Initiation Protocol Version 2 (RFC 3261, 3262, 3263, 3264) • SIP Proxy Redundancy - Dynamic via DNS SRV, A Records • Re-registration with Primary SIP Proxy Server • SIP Support in Network Address Translation Networks - NAT (including STUN) • SIPFrag (RFC 3420) • Secure (Encrypted) Calling via Pre-Standard Implementation of Secure RTP • Codec Name Assignment • Voice Algorithms: - G.711 (A-law and μ-law) - G.726 (16/24/32/40 kbps) - G.729 A - G.723.1 (6.3 kbps, 5.3 kbps) • Dynamic Payload Support • Adjustable Audio Frames Per Packet • DTMF: In-band and Out-of-Band (RFC 2833) (SIP INFO) • Flexible Dial Plan Support with Inter-Digit Timers • IP Address / URI Dialing Support • Call Progress Tone Generation • Jitter Buffer - Adaptive • Frame Loss Concealment • VAD - Voice Activity Detection with Silence Suppression • Attenuation / Gain Adjustments • MWI - Message Waiting Indicator Tones • VMWI - Voice Mail Waiting Indicator - Via NOTIFY, SUBSCRIBE • Caller ID Support (Name and Number) • Third Party Call Control (RFC 3725)
Provisioning, Administration & Maintenance	• Integrated Web Server Provides Web Based Administration and Configuration • Telephone Key Pad Configuration via Display Menu / Navigation • Automated Provisioning and Upgrade via HTTPS, HTTP, TFTP • Asynchronous Notification of Upgrade Availability via NOTIFY • Non-intrusive, In-Service Upgrades • Report Generation and Event Logging • Statistics Transmitted in BYE Message • Syslog and Debug Server Records - Configurable Per Line
Power Supply	• Power supply is optional and is purchased separately -- Models: PA100-NA, PA100-EU, PA100-UK, PA100-AU • Switching Type (100-240v) Automatic • DC Input Voltage: +5 Volts DC at 2.0 Amps Maximum • Power Consumption: 5 Watts • Power Adapter: 100-240v - 50-60Hz (26-34VA) AC Input, 1.8m cord
Physical Interface	• 2 100baseT RJ-45 Ethernet Ports (IEEE 802.3) • Handset: RJ-7 Connector • Built-in Speakerphone and Microphone • Headset 2.5 mm Port
LEDs	• Four (4) Call Appearance/Line Buttons with Associated Tricolor LED • Line LED State Indication: Active, Idle, On Hold, Unregistered • Speakerphone On/Off Button with LED • Headset On/Off Button with LED • Mute Button with LED • Message Waiting Indicator LED • Voicemail Message Retrieval Button • Hold Button • LED Test Function
Regulatory Compliance	FCC (Part 15, Class B) , CE, A-Tick, ICES-003
Security	• Password Protected System, Preset to Factory Default • Password Protected Access to Administrator and User Level Features • HTTPS with Factory Installed Client Certificate • HTTP Digest - Encrypted Authentication via MD5 (RFC 1321) • Up to 256-bit AES Encryption
Documentation	• Quick-Start Installation and Configuration Guide • User Guide • Administration Guide • Provisioning Guide - For Service Providers Only
Hardware
	• Pixel Based Display: 128x64 Monochrome LCD Graphical Display • Four Illuminated Call Appearance Line Buttons with Tricolor LED's • LED Indicates Line State – Active, Idle, On-Hold, Unregistered • Line LED Configurable to 13 Different States (On/Off, Color, Flash) • Dedicated Illuminated Buttons for: Audio Mute On/Off, Headset On/Off, Speakerphone On/Off, • Four Soft Key Buttons • Four Way Rocking Directional Knob for Menu Navigation • Voice Mail Message Waiting Indicator Light • Voice Mail Message Retrieval Button • Dedicated Hold Button • Settings Button for Access to Feature, Set-up, and Configuration Menus • Volume Control Rocking Up/Down Knob Controls Handset, Headset, Speaker, Ringer • Standard 12-Button Dialing Pad • High Quality Handset and Cradle • Built-In High Quality Microphone and Speaker • Headset Jack – 2.5 millimeter • LED Test Function • Two Ethernet LAN – 100BaseT RJ-45 • 802.3af Compliant Power over Ethernet (PoE) • Optional 5 volt DC Universal (100-240 Volt) Switching Power Adaptor - Power Supply is Ordered Separately
Environmental
Dimensions	7.68 x 6.30. x 7.09 in
WxHxD	(195 x 160 x 180 mm))
Unit Weight	2.15 lbs ( 0.9752 kg)
Operating Temperature	32º~113ºF (0º~45ºC)
Storage Temperature	-13º~185ºF (-25º~85ºC)
Operating Humidity	10~90% Non-condensing
Storage Humidity	10~90% Non-Condensing

Linksys-Cisco SPA400

Internet Telephony Gateway with 4 FXO Ports

Advanced Multi-Port PSTN Gateway Solution for the Linksys Voice System

* Functions as an Analog Line Gateway for a Linksys Voice System VoIP Network

* Integrated Voicemail Application Server with up to 32 Voicemail Accounts

* Perfectly Suited to Connect up to 4 Analog Lines

* Enables Linksys Voice System Users to Leave and Playback Voicemail Messages

* Designed only to be deployed with the Linksys SPA9000 (but can be used with Asterisk)

The SPA400 features the ability to connect up to four (4) standard analog telephones lines to a Linksys Voice System (LVS) VoIP network and includes the additional benefit of an integrated voicemail application. The SPA400 utilizes multiple analog phone lines and automatically routes calls to and from your existing PSTN telephone service. Designed to be implemented with the LVS IP Telephony System, the SPA400 enables cost-conscience business users to leverage the high value features generally found only on more expensive PBX equipment.

The SPA400 also includes an integrated voicemail application supporting up to 32 voicemail accounts or boxes with customized greetings, providing LVS users the ability to receive and playback voicemail messages from their Linksys IP Phones.

The SPA400 supports four (4) RJ-11 FXO ports to connect to the PSTN and also includes one 10/100 BaseT RJ-45 Ethernet interface that is used to connect to the local IP network. Compact in design, the SPA400 is perfectly suited for a small business in providing reliable operation for both VoIP and PSTN voice communication service.

Installed by the service provider, reseller, or end user, the SPA400 enables transparent PSTN gateway functionality for a VoIP telephony network and is a key element in building a solid business communications system.

VIP-1680 series VoIP Gateway

The PLANET VIP-1680 series VoIP Gateway are dual standard (SIP and H.323) compliant residential gateways that provide a total solution for integrating voice-data network and the Public Switched Telephone Network (PSTN). The VIP-1680 series not only provide high-quality voice communications but also offer secure, reliable Internet sharing capabilities for daily voice and Internet communications.

With advanced DSP processor (TI) and cutting-edge VoIP technology, the PLANET VIP-1680 series are capable of handling both SIP and the H.323 calls. With up to 16 registrations to the SIP proxy or H.323 Gatekeeper and allows 16 lines analog voice and fax communication over a traditional data communications/data networking digital Internet, the VIP-1680 series are able to make calls to either H.323 or SIP voice communication environment. The VIP-1680 series are equipped with a Ethernet switch and built-in NAT router function that provides Internet access using only one IP address; with these features, users can now enjoy high-quality voice calls and secure Internet access without interfering with routine activities.

Meanwhile, the PLANET VIP-1680 series are designed in comfort and ease-of-use for customers. The VIP-1680 series not only inherit traditions of quality voice communications and real-time fax data over IP networks but also eliminate the human resource in VoIP network deployment. With optimized H.323/SIP architecture, the PLANET VIP-1680 series are ideal choices for P2P voice chat, ITSP cost-saving solution, as well as providing network-converting feature to translate the packet network into traditional PBX system.

With built-in PPPoE/DHCP/DDNS clients, up to 16 concurrent connections can be offered in VIP-1680 series, therefore voice communications can be established anywhere around the world. The PLANET VIP-1680 series come with intuitive user-friendly yet powerful management interface (web/telnet/console) that can dramatically reduce IT personnel resource and complete VoIP deployment in a short time. With the remote management capability provided, the VoIP administrators can monitor machine/network status, or proceeds maintenance/trouble-shooting service via Internet browser or telnet session. Besides, the VIP-1680 series provide voice channels status display and optimized packet voice streaming over managed and public (Internet) IP networks.

VIP-1680	H.323 / SIP VoIP Gateway (1 x WAN, 1 x LAN, 8 *FXO/8 * FXS; life line support)
VIP-1680FO	H.323 / SIP VoIP Gateway (1 x WAN, 1 x LAN, 16 * FXO)
VIP-1680FS	H.323 / SIP VoIP Gateway (1 x WAN, 1 x LAN, 16 * FXS)