Linksys SPA-2002 2 Port FXS Analog VoIP Adapter                 $78.95     The Linksys SPA-2002 features two POTS (Plain Old Telephone Service) ports for connection to existing analog phones, fax machines, PBX and key system communication platforms. The SPA includes an Ethernet interface for connection to a home or office LAN. Each SPA service line can be independently configured via software controlled by the service provider and/or the end user. The SPA-2002 uses the SIP Protocol, and is compatible with SIP Version 2. The Linksys SPA-2002 features the new ID that Linksys has rolled out on the SPA-1001 and SPA-2100. From a feature and functionality point of view, the SPA-2002 ATA is identical to the SPA-2000. The SPA-2002 hardware design provides expanded memory support (twice the memory of the SPA-2000) to accommodate new features available in future SPA-2002 firmware updates. Linksys SPA-2002 Features: Terminating Impedance Agnostic - 8 Settings Call Waiting, Cancel Call Waiting Caller ID with Name / Number Caller ID Blocking Call Waiting Caller ID with Name / Number Call Forwarding: No Answer / Busy / All Do Not Disturb Call Transfer Three-Way Conference Calling with Local Mixing Message Waiting Indication - Visual and Tone Based Call Return Call Back on Busy Call Blocking with Toll Restriction Delayed Disconnect Distinctive Ringing Off-Hook Warning Tone Selective / Anonymous Call Rejection Hot Line and Warm Line Calling Speed Dialing of 8 Numbers / Addresses Music On Hold

Linksys PAP2T-NA Analog VoIP Adapter (2) FXS Unlocked           $82.95

       T38 Fax Compatible

       

The Linksys PAP2T Internet Phone Adapter enables high-quality feature-rich VoIP (voice over IP) service through your broadband Internet connection. Just plug it into your home Router or Gateway and use the two standard telephone ports to connect analogue phones or use one of the ports for a fax machine.

Download PAP2T-NA DataSheet

Linksys PAP2T-NA Analog VoIP Adapter (2) FXS Unlocked

Each phone port operates independently, with separate phone service and phone numbers -- like having two telephone lines. You'll get clear reception and a reliable fax connection, even while using the Internet at the same time.

With Internet telephony, along with low domestic and international phone rates, an impressive array of special telephone features are available. Choose your preferred free local dialing US area code, regardless of where you live. Or add a virtual telephone number in any area code, forwarded to your Internet phone. You can even add a toll-free number.

The Linksys Internet Phone Adapter is compatible with these and all of the other special telephone features that are available from your Internet telephony service provider, such as Caller ID, Call Waiting, Voicemail, Call Forwarding, Distinctive Ring, and much more.

Let the Linksys Internet Phone Adapter enhance your existing Internet connection with a high-quality high-value VoIP service.

Features

• Two voice ports (RJ11) for analog phones or Fax machines with two independent telephone numbers
• One RJ-45 port for 10/100 Mbps Ethernet connection
• Supports Dynamic Host Configuration Protocol (DHCP)
• Supports Session Initiation Protocol (SIP)
• Supports multiple voice compression standards: G.711, G.726, G.729, and G.723.1
• Web-based configuration through a built-in web server
• Supports DTMF tone detection and generation
• Supports FSK Caller ID, DTMF Caller ID and FSK VMWI
• Supports Echo Cancellation and Voice Activity Detection (VAD)
• Password protected access and configuration
• Supports auto-provisioning with remote firmware upgrade

Linksys SPA3102 NA 1FXS / 1FXO Analog VoIP Gateway       $94.95 

The SPA3102 NA from Linksys is slated as the replacement for the popular SPA3000, and offers both 1 FXS station side port and 1 FXO PSTN jack, and features integrated call routing for local and emergency calls, excellent for service providers and remote offices!

Phone Adapter + PSTN Gateway

Linksys SPA3102 NA Phone Adapter + PSTN Gateway The SPA3102 NA features VoIP adapter functionality found in the SPA2002 and SPA1001 with the additional benefit of an integral connection for legacy telephone network "hop-on, hop-off" applications. SPA3102 NA users will be able to leverage their broadband phone service connections more than ever by automatically routing local calls from cell phones and land lines to a VoIP service provider and vice versa.

A typical user calling from a land line or mobile phone will be able to reduce and even eliminate international and long distance telephone charges by first calling their SPA3000 via a local phone number or by using a telephone connected directly to the unit. The advanced authentication and call routing intelligence programmed into the SPA-3102 NA will connect the caller via the Internet to the far end destination with security and ease. Using the SPA3102 NA at the far end, calls can be answered immediately or further processed as a local call to any legacy land line or mobile phone allowed by the SPA-3102 NA dial plan.

If power is lost to the unit or the VoIP service is down, calls can be sent to a traditional carrier via the FXO interface

Linksys SPA2100 Analog Telephone Adaptor                               $87.95

      

All new Linksys SPA2100 Dual FXS SIP Adapter. Upgrade from SPA2000 adding a second Ethernet (RJ45) port and additional feautures.

Inexpensive, easy-to-install and simple-to-use, the Linksys SPA2100 connects standard telephones and fax machines to IP-based data networks. IP telephony service providers and enterprise users can offer traditional and enhanced communication services via the customers' broadband connection to the Internet or Local Area Network (LAN)

The SPA2100 features two POTS (Plain Old Telephone Service) ports for connection to existing analog telephones. The SPA2100 includes an Ethernet interface for connecting to a home or office PC (LAN) as well as an Ethernet connection to the broadband modem or router (WAN). Each SPA-2100 service line can be independently configured via software controlled by the service provider and/or the end user. 

With the SPA2100, individuals and companies are able to protect and extend their past investments in telephones, conference speakerphones, and fax machines as well as control their migration to IP with an extremely affordable, incremental investment.

Linksys SPA2100 Key Features

• Terminating Impedance Agnostic - 8 Settings
• Call Waiting, Cancel Call Waiting
• Caller ID with Name / Number
• Caller ID Blocking
• Call Waiting Caller ID with Name / Number
• Call Forwarding: No Answer / Busy / All
• Do Not Disturb
• Call Transfer (Blind and Consultative)
• Three-Way Conference Calling with Local Mixing
• Message Waiting Indication - Visual and Tone Based
• Call Return
• Call Back on Busy
• Call Blocking with Toll Restriction
• Delayed Disconnect
• Distinctive Ringing - Calling and Called Numbers
• Off-Hook Warning Tone
• Selective / Anonymous Call Rejection
• Hot Line and Warm Line Calling
• Speed Dialing of 8 Numbers
• Music on Hold
• Fax - G.711 Pass-Through or Real Time Fax over IP via T.38 (Pending)

Grandstream Handytone 286
(1) FXS Port - Analog Adapter                                                              $49.95       

Grandstream HandyTone 286 ATA-286 is an award-winning next generation VoIP analog telephone adapter based on industry open standards. Built upon innovative technology, Grandstream HandyTone ATA-286 features market leading superb sound quality, compact size, and rich functionalities at highly-affordable price.

        

             

• Support SIP 2.0, TCP/UDP/IP, RTP/RTCP, HTTP, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP protocols
• Support NAT traversal using IETF STUN and symmetric RTP (compatible with Cisco’s ATA-186, etc)
• Advanced Digital Signal Processing (DSP) to ensure superb hi-fidelity audio quality
• Provides 1 LAN port and 1 FXS interface for any analog telephones, cordless phones, and fax machines
• Support transparent Fax pass-through and in the future T.38 (pending)
• Interoperable with various 3rd party SIP end user device, Proxy/Registrar/Server, and gateway products
• Advanced and patent pending adaptive jitter buffer control, packet delay and loss concealment technology
• Support popular codecs including G.723.1 (5.3K/6.3K), G.729A/B, G.711 (a-law and u-law), G.726, and G.728. Dynamic negotiation of codec and voice payload length
• Support standard voice features such as Caller ID Display or Block, FLASH, in-band and out-of-band DTMF (RFC2833), Dial Plans, off-hook auto dial, early dial, click-to-dial
• Support acoustic echo cancellation, voice mail with indicator, downloadable ring tone (pending)
• Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control)
• Support DIGEST authentication and encryption using MD5 and MD5-sess.
• Provide easy configuration thru manual operation (attached analog phone keypad and voice prompt, Web interface) or personalized automated provisioning via central configuration file for mass deployment.
• Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)
• NAT-friendly remote software upgrade capability (via tftp) even from behind firewalls/NATs.
• Support for fail-over SIP server and DNS server (pending)

Grandstream Handytone GS-486 Multiport Adapter                   $59.95         

Grandstream HandyTone 486 is an “All-In-1” VoIP IAD based on SIP standard. Built upon Grandstream's innovative technology, HandyTone 486 features superb sound quality, rich functions, high degree of integration, ease of use, compact size, and ultra-affordability.

Key Features

• Supports SIP2.0, TCP/UDP/IP, RTP/RTCP, HTTP, ICMP, ARP/RARP, DNS, DHCP (both client and server), NTP, PPPoE, TFTP, etc.
• Built-in router, NAT and Gateway.
• Powerful digital signal processing (DSP) to ensure superb audio quality; advanced adaptive jitter control and packet loss concealment technology.
• Ultra compact (wallet size) and lightweight design, great companion for travelers.
• Supports various vocoders including G.711 A-/U-law, G.723.1, G.729A/B, G.728, G.726, iLBC.
• Supports Caller ID/Name display or block, Hold, Call Waiting, Call Transfer, Call Forward, Flash, Three-Way Conferencing (pending).
• Supports customizable ring tone (pending).
• Fax pass through and T.38 (pending).
• Supports silence suppression and VAD, AGC, and echo cancellation (G.168).
• Supports standard encryption and authentication (DIGEST using MD5 and MD5-sess).
• Supports layer-2 (802.1Q VLAN, 802.1p) and layer-3 (DiffServ, ToS) QoS.
• Supports automated NAT traversal without manual manipulation of firewall/NAT.
• Supports remote automated provisioning and software upgrade even through firewall/NAT to enable “zero configuration” and “plug-and-dial” for end users.
• Supports device configuration via built-in IVR, Web browser or central configuration file.
• Supports DNS SRV Look Up.
• SIP Server Fail Over (pending).

NEW Grandstream Handytone 488 Multiport Analog Adapter
SIP VoIP FXS + FXO Gateway                                                             $69.95

Grandstream HandyTone 488 is an “All-In-1” VoIP IAD based on SIPstandard. Built upon Grandstream’s innovative technology，HandyTone 488 features superb sound quality, rich functions, high degree of integration, ease of use, compact size, and ultra-affordability.

                 

• Dual RJ45 (LAN & WAN)
• Dual RJ11 - (1) FXS, (1) FXO
• Built in DHCP, NAT & Router
• PSTN Pass-through

   

• Support SIP2.0 ， TCP/UDP/IP ， RTP/RTCP ， HTTP ， ICMP ， ARP/RARP ， DNS ， DHCP (both client and server) ， NTP ， PPPoE，TFTP, etc.
• Built-in router, NAT and Gateway
• Powerful digital signal processing (DSP) to ensure superb audio quality; advanced adaptive jitter control and packet loss concealment technology
• Ultra compact (wallet size) and lightweight design, great companion for travelers
• Support various vocoders including G.711 A-/U-law ， G.723.1, G.729A/B, G.728, G.726, iLBC.
• Support Caller ID/Name display or block, Hold, Call Waiting, Call Transfer, Call Forward, Auto Answer (Intercom), Flash
• Support customizable ring tone and fax pass through
• Support silence suppression and VAD ， AGC ， and echo cancellation (G.168)
• Support standard encryption and authentication (DIGEST using MD5 and MD5-sess)
• Support layer-2 (802.1Q VLAN, 802.1p) and layer-3 (DiffServ, ToS) QoS
• Support automated NAT traversal without manual manipulation of firewall/NAT
• Support remote automated provisioning and software upgrade even through firewall/NAT to enable “zero configuration” and “plug-and-dial” for end users
• Support device configuration via built-in IVR, Web browser or central configuration file

Grandstream Budgetone GS-101

The Grandstream Budgetone GS-101 IP Phone is an award-winning next generation IP network telephone based on industry open standards. Built upon innovative technology, Grandstream IP Phone features market leading superb sound quality and rich functionalities at ultra-affordable price.

Grandstream Budgetone GS-101 VoIP Business Phone SIP                          $54.95

               

Grandstream GS-101 Features:

• Support SARP/RARP, ICMP, DNS, DHCP, NTP, TFTP protocols
• Support NAT traversal via STUN & symmetric RTP
• Interoperable with various 3rd party SIP end user device, Proxy/Registrar/Server, and gateway products
• Advanced Digital Signal Processing (DSP) to ensure superb hi-fidelity audio quality
• Advanced and patent pending adaptive jitter buffer control, packet delay & loss concealment technology
• Support popular vocoders including G.723.1 (5.3K/6.3K), G.729A/B, G.711 (a-law and u-law), G.726, G.728, and wide-band G.722 (Model 102D). Dynamic negotiation of codec and voice payload length
• Support standard voice features such as Caller ID Display or Block, Call Waiting, Hold, TransfForward, FLASH, in-band and out-of-band DTMF (RFC2833), Dial Plans, off-hook auto dial, configurable emergency dialing (e.g., 911), early dial, click-to-dial
• Support 3-way conferencing (Model 102D), full duplex hands-fredomain acoustic echo cancellation (pending), redial, call log, volume control, voice mail with indicator, downloadable ring tone (pending)
• Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control)
• Support DIGEST authentication and encryption using MD5 and MD5-sess.
• Provide easy configuration thru manual operation (phone keypad and Web interface) or personalized automated provisioning via central configuration file for mass deployment.
• Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)
• NAT-friendly remote software upgrade capability (via tftp) even from behind firewalls/NATs.
• Support for fail-over SIP server and DNS server (pending)

The Grandstream GS-101 has one Ethernet port.

Grandstream Budgetone GS-102                                                         $59.95

The Grandstream Budgetone GS-102 IP Phone is an award-winning next generation IP network telephone based on industry open standards. Built upon innovative technology, Grandstream IP Phone features market leading superb sound quality and rich functionalities at ultra-affordable price.

                

Grandstream GS-102 Features:

• Support SARP/RARP, ICMP, DNS, DHCP, NTP, TFTP protocols
• Support NAT traversal via STUN & symmetric RTP
• Interoperable with various 3rd party SIP end user device, Proxy/Registrar/Server, and gateway products
• Advanced Digital Signal Processing (DSP) to ensure superb hi-fidelity audio quality
• Advanced and patent pending adaptive jitter buffer control, packet delay & loss concealment technology
• Support popular vocoders including G.723.1 (5.3K/6.3K), G.729A/B, G.711 (a-law and u-law), G.726, G.728. Dynamic negotiation of codec and voice payload length
• Support standard voice features such as Caller ID Display or Block, Call Waiting, Hold, TransfForward, FLASH, in-band and out-of-band DTMF (RFC2833), Dial Plans, off-hook auto dial, configurable emergency dialing (e.g., 911), early dial, click-to-dial
• Support 3-way conferencing (Model 102D), full duplex hands-fredomain acoustic echo cancellation (pending), redial, call log, volume control, voice mail with indicator, downloadable ring tone (pending)
• Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control)
• Support DIGEST authentication and encryption using MD5 and MD5-sess.
• Provide easy configuration thru manual operation (phone keypad and Web interface) or personalized automated provisioning via central configuration file for mass deployment.
• Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)
• NAT-friendly remote software upgrade capability (via tftp) even from behind firewalls/NATs.
• Support for fail-over SIP server and DNS server (pending) The Grandstream GS-102 has two Ethernet ports.

Linksys SPA841 2-4 Line SIP Phone                                                   $64.95

                      

The Linksys SPA841 business phone is the premier entrant in an IP telephone product line from Sipura Technology that leverages the global success of Sipura and the rapid adoption of VoIP technology in worldwide VoIP deployments.

Linksys SPA 841 SIP VoIP Phone

The Linksys SPA841 comes standard with 2 line appearances and can be upgraded to 4 line appearances for an additional $30.00 

The Sipura SPA841 IP telephone can be configured as a two (2) line or, via a simple software upgrade, a four (4) line full featured business phone with pixel based graphical display, speakerphone and headset port. Stylish and functional in design, the SPA-841 can be used in residential, SOHO, enterprise and small to medium business service offerings including IP PBX, hosted IP telephony and IP Centrex. The SPA841 leverages Sipuras market leading technology and manufacturing proficiency to deliver an upgradeable, high quality IP telephone unparalleled in value and support.

Interoperability and SIP Based Feature Set

Experienced telephony service network operators recognize that technical acumen coupled with responsive pre and post sales support are critical for a successful deployment. Sipuras extensive interoperability track record with VoIP industry infrastructure leaders via standards based and platform specific SIP signaling enable network providers to quickly roll-out competitive, feature rich service offerings. Armed with a mature feature set with hundreds of programmable parameters, the SPA-841 utilizes the call processing functionality found in existing Sipura products. Sipura VoIP endpoint solutions solve many time-to-market requirements of enterprise users and leverage the advantages of an IP network like easy acceptance of station moves, presence and shared line appearances across geographically separate locations

Linksys SPA941 SIP Phone                                                           $149.95

                  

Stylish and functional in design, the Linksys SPA941 VoIP Phone is ideal for a residence or business using a hosted IP telephony service, an IP PBX, or a large scale IP Centrex deployment. The SPA941 leverages industry leading VoIP technology from Linksys to deliver an upgradeable high quality IP Phone that is unparalleled in features, value, and support.

Linksys SPA941 SIP screenphone with (2) Line appearances, expandable to (4)

Linksys SPA941 Standard features on the Linksys SPA941 include two active lines, a high resolution graphical display, speakerphone, and a 2.5 mm head-set port. With a simple software update, the Linksys SPA941 is upgradeable to a four line phone. 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.

Linksys SPA941 Comprehensive Interoperability and SIP Based Feature Set

Based on the SIP standard, the Linksys SPA941 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 SPA941 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 Linksys SPA941.

Carrier-Grade Security, Provisioning, and Management
The SPA941 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).

Linksys SPA941 Hardware Features:

• (1) RJ-45 10BaseT Ethernet Port
• High Quality, "Cisco/Linksys?" Distinctive ID
• Hi-Resolution Pixel Based Display
• Larger Display vs. SPA841
• 128 x 64 vs. 128 x 48
• Four Line Keys
• Four Soft Keys
• Solid Handset
• Headset Port 2.5mm
• Full Duplex Speakerphone

All SW Features of SPA84x Including . . .
 

• Interoperability with Asterisk and other SIP based platforms
• Excellent Voice Quality
• Secure Calling via sRTP
• Network Based Ring Tone Support
• Call Transfer, DND, Conferencing, Call FWD, etc.
• SIP B and Bridged Line Appearance Support
• Remote Provisioning via HTTPS, HTTP, TFTP

Linksys WIP300 Wi-Fi IP Phone 802.11b/g w/ Color LCD                 $229.95

The Linksys WIP300-NA WiFi Phone enables high-quality voice over IP (VoIP) service through a Wireless-G network and high-speed Internet connection. Connect at home, your office, or at a public hotspot, and make low-cost phone calls through your Internet Telephony Service Provider. The WIP300 WiFi phone operates in the 2.4GHz band, supports IEEE802.11 b/g and the latest VoIP SIP protocol.

  

Linksys WIP300-NA SIP V2.0 Handset - Supports WIFI 802.11b / g, Color LCD, POP3, SMTP

Download Linksys WIP300 Data Sheet
Download Linksys WIP300 User Guide

The WIP300 Wireless-G IP Phone operates in the 2.4GHz band, supports 802.11g and the latest VoIP SIP protocols. The large, full-color high resolution display features an intuitive user interface enabling users to easily and quickly configure the handset using Secure Easy Setup (SES).

SES is a fast, secure way of connecting the WIP300 Wireless-G IP Phone to a Linksys Wireless-G broadband router. Just press the associated SES button on both devices and your connection will automatically be configured and secured with a custom SSID using powerful WPA (Wi-Fi Protected Access) encryption. That’s it! Within seconds you have successfully set up a secure connection.

Handset features include caller ID, call forwarding, call transfer, stores call history and can save 200 phone book entries. Personalize your phone with a selection of ringtones and choose different wallpaper images that reflects your style.

Get the value of low-cost VoIP service with the convenience of Wireless-G connectivity with the Wireless-G IP Phone from Linksys.

Linksys WIP300 Additional Features include:

• Pixel-based display—Provides intuitive access to calling features
• Nine speed dials configurable in the set
• Comfort noise generation (CNG), voice activity detection (VAD), adaptive jitter buffer, and echo cancellation
• RF and battery level indication
• Local phone book
• Embedded 2.4GHz antenna
• ABS+PC plastic housing
• 1.8” COLOR TFT LCD with backlight
• Simple keypad with backlight
• Remote Firmware upgrading via WiFi
• SIP v2 signaling protocol, RFC-3261
• POP3/SMTP E-mail access (optional)
• SMS (optional, by system default)
• USB charger interface

UTStarCom F1000 G WIFI VOIP Phone WLAN 802.11B/G                   134.95

    

UTStarcom F1000 WIFI Mobile Handset is SIP based WLAN phone that is easily integrated with most SIP based IP PBX systems.

UTStarcom F1000 G WIFI VOIP Phone WLAN Wireless 802.11B

The latest version of the UTStarcom F1000 handset supports 802.11G in addition to 802.11 B.

UTStarcom F1000 Key Features:

• Bar Type Design
• 802.11b/g Wireless
• SIP (Session Initiation Protocol) Protocol
• DHCP
• 3-Way Calling
• Call Waiting
• Call Transfer
• Call Forwarding
• Call Hold / Resume
• 802.1X Authentication

UTStarcom F1000 Voice Specifications

• Codec G.711, G.729a/b
• Frequency Band 2.4GHz
• Transmission Output 20mW
• Comfort Noise Generation (CNG)
• Voice Activity Detection (VAD)
• Adaptive Jitter Buffer
• Echo Cancellation

UTStarcom F1000 IP Protocol

• RTP (Real-Time Transfer Protocol) RFC 1889 / RTCP FC 1890
• SDP (Session Description Protocol) RFC 2327
• SAP (Session Announcement Protocol)
• SIP RFC 3261, 3264, and 3515
• DHCP
• DTMF RFC 2833
• TFTP
• 802.1X Authentication
• PPPoE Authentication
• 64 and 128bit Wired Equivalent Privacy (WEP)

UTStarcom F1000 Technical

• Weight 111g
• Charger Input 100~240VAC: 50~60 Hz 120mA
• 4 Hour Talk Time

• Standby Time - 30-50 Hours