Sip Phone Gateway

Panasonic KX-NCS4104 4 Channel H.323 / SIP Gateway Activation Key $184.99 "Panasonic KX-NCS4104, The Panasonic KX-NCS4104 is a 4 channel H.323 / SIP gateway activation key. KX-NCS4104 Features: 4 Channel H.323 / SIP Gateway Activation Key, Will Not Work Without Phone System: Panasonic KX-TDE100 or KX-TDE200"

3Com Sip Phone 655005001 $93.25 3Com Sip Phone 655005001

Panasonic / BASIC SIP PHONE / UT113-B $108.29 Panasonic - BASIC SIP PHONE - UT113-B

Panasonic / BASIC SIP PHONE / UT123-B $130.46 Panasonic - BASIC SIP PHONE - UT123-B

Panasonic / STANDARD SIP PHONE / UT136-B $226.98 Panasonic - STANDARD SIP PHONE - UT136-B

Panasonic / STANDARD SIP PHONE / UT133-B $194.81 Panasonic - STANDARD SIP PHONE - UT133-B

3COM Sip Phone 655005001 3CI1002 $93.25 3COM Sip Phone 655005001 3CI1002

Panasonic Executive Sip Phone KXUT670 $460.25 Panasonic Executive Sip Phone KXUT670

Panasonic Standard Sip Phone KXUT136B $218.75 Panasonic Standard Sip Phone KXUT136B

Aastra MBU400 SIP DECT 6.0 Phone $254.99 "Aastra MBU 400 Brand New Includes One Year Warranty, Item # A1762-0000-02-00 The Aastra MBU 400 SIP DECT 6.0 phone is an affordable, scalable multi-handset SIP DECT mobility solution for small business. The MBU 400 DECT gateway supports eight unique SIP registrations and one FXO port providing a fully integrated SIP and Analog service for small business. It can also be provisioned and managed via the AastraLink Pro 160 web UI, providing a single tool for configuration and maintenance of the handsets and other Aastra SIP terminals. Employing DECT 6.0 technology, the MBU 400 offers virtually interference-free performance in any work or residential environment. The handset is equipped with a 1.5"" backlit color graphical display providing caller ID, call waiting, DND, transfer, hold and a variety of information at-glance. This stylish handset also offers features such as speakerphone, intercom between handsets, polyphonic ring tones and a large directory for up to 170 entries. MBU 400 Features: DECT 6.0 Technology (1.9GHz) Interference-Free & Wide Range 30 more Battery life, Caller ID / Call Waiting, Expandable Up To 8 Handsets, Hearing Aid Compatible, Acoustic Echo Cancellation, Speakerphone, 1.5"" Full Color LCD Display 65k High Resolution, Adjustable Screen Contrast, Backlit Keypad, 30 Station Name / Number Caller ID Memory, 170 Station Phone Directory / Dialer, Call Forwarding, Missed Call Indicator, 9 Selectable Polyphonic Ringtones, Call Transfer, Redial, Hold, Date / Time Display, Do Not Disturb, Up to 12 Hours Talk Time, Up To 200 Hours Standby Time, Low Battery / Out of Range Indicator, Headset Jack, Belt Clip - Included w/ Brand New Models, Multi-Language Menus, Base Unit Features: New DECT 6.0 Technology -1.9GHz SIP DECT Cordless Base, 1 FXO Port & 1 LAN Port, Multiple Handset Capability: - Support For 8 Paired 420d DECT Handsets Each w/ A Unique VoIP Telephone Number, Centralized/Shared Phonebook For All Paired Handsets, Embedded HTTP Web-server For Configuration, Integrated Antennas, 50m Indoor & 300m Outdoor Range Additional Handsets For This Model Are: span class="

Panasonic / SIP DECT Phone Corded / KX-TGP551T04 $236.58 Panasonic - SIP DECT Phone Corded - KX-TGP551T04

Panasonic / SIP IP DECT CORDLESS PHONE / TGP550T04 $237.69 Panasonic - SIP IP DECT CORDLESS PHONE - TGP550T04

Snom Meeting Point Sip Conference Phone Sipconference $632.75 Snom Meeting Point Sip Conference Phone Sipconference

Pingtel Xpress VOIP/SIP/IP Phone 960003907 $93.25 Pingtel Xpress VOIP/SIP/IP Phone 960003907

Panasonic Executive Sip Series Corded Phone KXUT248B $303.85 Panasonic Executive Sip Series Corded Phone KXUT248B

Panasonic Sip Office Desk Phone KXUT133B $187.13 Panasonic Sip Office Desk Phone KXUT133B

Panasonic KX-NCS4102 2 Channel H.323 / SIP Gateway Activation Key $117.99 "Panasonic KX-NCS4102, The Panasonic KX-NCS4102 is a 2 channel H.323 / SIP gateway activation key. KX-NCS4102 Features: 2 Channel H.323 / SIP Gateway Activation Key"

Cisco Unified IP Phone 7965G VoIP Phone Sccp Sip CP7965G $353.38 Cisco Unified IP Phone 7965G VoIP Phone Sccp Sip CP7965G

Cisco Unified IP Phone 7945G VoIP Phone Sccp Sip CP7945G $285.85 Cisco Unified IP Phone 7945G VoIP Phone Sccp Sip CP7945G

Cisco Unified IP Phone 7906G VoIP Phone Sccp Sip CP7906G $123.6 Cisco Unified IP Phone 7906G VoIP Phone Sccp Sip CP7906G

Obihai OBI110 VoIP Telephone Adapter with Google Voice, SIP & Phone Line Gateway $49.99 Type: Accessories Security: Local Access Interface: IVR Password Remote Access Interface: User Name & Password Access via HTTP, TFTP - HTTPS Device Web Page Standard: HTTP v1.1, XMLv1.0 Secure Remote Provisioning: TFTP, HTTP, HTTPS Ports: Internet (WAN): 1 x 10/100BaseT Ethernet Port (802.3) Phone (FXS): 1 x RJ-11 FXS Analog Phone Port Line (FXO): 1 x RJ-11 FXO Analog Line Port LEDs: Power On, Device Status, Upgrade in Progress Status, Packet RX/TX, Phone Port Status (Enabled, In-Use), Line Port Status(Enabled, In-Use) Power: Input Power: AC Input: 100 to 240 Volts 0.3A 50-60Hz (26-34 VA) Output Power: DC: +12V 1.0 Amp Max Dimensions: 4.5" x 4.2" x 1.2" Weight: 9 ounces Temperature: 0ºC to 45ºC

Panasonic KX-HGT100 SIP Phone $109.99 "Panasonic KX-HGT100 Brand New Includes One Year Warranty, The Panasonic KX-HGT100 uses the latest SIP (Session Initiation Protocol) technology, helping to lower business telephone costs and simplify communication management. It supports local office users or remote workers, connecting over high-speed broadband IP network from virtually anywhere. The KX-HGT100 is compatible with the KX-TDE and KX-NCP PBX Panasonic Communications Systems so its ideal for companies with geographically diverse office locations and helps connect all employees with customers, wherever they may be. KX-HGT100 Features: Program Key, Hybrid System Corded IP Phone, SIP Voice Over Internet Protocol (VoIP), Caller ID, Voicemail Compatibility, Speakerphone, 2-Line / 16-Character LCD Display, Speed Dial, Call Transfer, Redial, Hold, Mute, IEEE 802.3af Power over Ethernet (PoE) Support, Optional AC Adapter (Not Included), Two-Port 10/100 Mbps Ethernet Switch, Headset Jack"

Panasonic® Basic SIP Phone. Each $101.74 Manufacturer: Panasonic. Each. Panasonic basic SIP phone Certified for Asterisk and Broadsoft Easy operation 3 Way conference call 100 entry phonebook Plug and play configuration 10/100-BaseTX Ethernet port, PoE Green { Large, clear LCD displays with easy

Panasonic Executive SIP Phone. Each $523.98 Manufacturer: Panasonic. Each. 6 SIP accounts 7 inch, color touch screen Integration with IP Cameras (up to 16) HD Voice (G.722) 100 entry phonebook 3-way conference call support 2 x GbE ports, PoE Full duplex speaker phone Built in Bluetooth Plug and Pl

Panasonic KX-NCS4716 16 Channel SIP-Phone License $444.99 "Panasonic KX-NCS4716, The Panasonic KX-NCS4716 is a 16-channel SIP extension activation key. KX-NCS4716 Features: 16-Channel SIP Extension Activation Key, Will Not Work Without Phone System: Panasonic KX-TDE100 or KX-TDE200"

4ch Sip Trunk Card $538.05 - Panasonic 4-channel SIP gateway card (SIP-GW4)- Compliant with RFC 3261, 3262, 3264, 3311, and 4028 protocols, and ITU-T G.729A/B and G.711 CODEC methodsKXTDA5450

DECT Cordless Phone with Base $183.44 - Snom VoIP wireless DECT phone- 3 Concurrent VoIP calls (SIP 2.0)- Up to 8 handsets- Call transfer (attended and blind); call hold SIP- Call log: incoming, outgoing and missed calls- Call swap- Intercom - handset to handset via gateway- Phonebook 170 ent

SIP Trunking $62.85 Since the mid-1990s IP telephony has become a widespread means of communication for businesses and service providers. Roughly three-quarters of large companies in the U.S. have already switched to IP telephony, enabling rich-media applications such as collaborative meetings, video, presence-based communication choices, rich hard phone or soft phone displays and end user call control mechanisms. Service provider backbone networks have also largely converted to VoIP transport realizing bandwidth and converged network architecture benefits. Yet TDM trunks are still the predominant mechanism to interconnect businesses with the PSTN (service provider), limiting the inter-business communications to the single-media (voice-only) transport of the traditional PSTN. To realize the promise of VoIP and enable rich-media business-to-business collaborative applications, service providers have in 2008 started offering implementable SIP trunk interconnects. Enterprise interest in SIP trunks for cost benefits, transport benefits as well as new productivity applications have also increased dramatically of late. SIP Trunks provides an overview of the trends and technologies in evolving PSTN interconnect from TDM to SIP-based transport. It discusses the real benefits and the popular myths surrounding SIP trunks and helps you evaluate what real benefits you could implement for your business. The book provides an in-depth discussion of planning your network for SIP trunk implementation and how to evaluate SIP trunk offerings. Practical guidance around RFP structure is given, including questions to ask the service provider, and providing a sample cost analysis. It also presents an in-depth discussion of how to deploy SIP trunk interconnects to an enterprise network. The possible deployment models are covered including the trade-offs and network design issues such as security considerations, call admission control and handling new call flows. It offers concrete implementation steps and realistic best practices to follow during the implementation. Network implementation is illustrated with a case study. The book concludes with an overview discussion of the future of unified communications networks and how business transformation due to end-to-end VoIP connectivity (of which SIP trunking is a critical piece) might evolve. This coverage helps the reader visualize how their business might transform over time and how best to plan and prepare their business to capitalize on the benefits.

Cisco 7940 IP Phone Sip And Mgcp Lic SWSMUL7940 $140.4 Cisco 7940 IP Phone Sip And Mgcp Lic SWSMUL7940

Cisco And 3PCC Sip And Mgcp License For Single 7960 IP Phone SWSMUL7960 $140.4 Cisco And 3PCC Sip And Mgcp License For Single 7960 IP Phone SWSMUL7960

Unified SIP Phone 3905 - VoIP phone - with 1 x User Connect Licensing (UCL) $63.99 Cisco Unified SIP Phone 3905 - VoIP phone - SIP - charcoal - with 1 x User Connect Licensing (UCL)

Zoom 5801 VoIP Gateway $59.99 1 1 x RJ-11 FXO 1 x RJ-11 FXS 1 x RJ-45 LAN 1.10" x 5.75" x 4.41" 5801 5801 VoIP Gateway 58010011 5801 VoIP Gateway Power adapter Quick Start Guide CD ROM Ethernet cable Phone cord Provisioning/configuration/Authentication Password-protected, Web based administration ARC4 Encryption for TFTP Configuration Profiles Authentication (Digest using MD5) Web based administration Layer 2 (802.1Q VLAN) Layer 2 Class-of-Service CoS) Tagging (802.1P) ICMP HTTP DHCP Voice over IP (VoIP) Protocol: SIPv2 Session Initiation Protocol (RFC 3261, 3262, 3263, 3264) SDP Session Description Protocol (RFC 2327) RTP Real Time Protocol (RFC 1889, 1890) RTCP Real-Time Control Protocol (RFC 1889) X-NSE Tone Events for SIP/RTP (RFC 2833) AVT Tone Events for SIP/RTP (RFC 2833) Power on Auto Registration Re-registration with SIP Proxy Server SIP over UDP SIP authentication (HHP Digest with MD5) The economical Zoom Model 5801 Voice over IP Telephone Adapter allow standard analog telephones to make and receive calls over a cable or ADSL Internet connection. Simply plug a conventional or cordless phone into the adapter and then plug the telephone adapter into a broadband connected router or Internet gateway. A minimum of five attached phones can be rung simultaneously using the built-in ring generator. External G.165 G.168 G.711 Phone Cable Twisted Pair VoIP Gateway Zoom Zoom Telephonics, Inc www.zoom.com

Cisco Unified Sip Phone 3911 VoIP Phone Sip Dark Gray With 1 X User License For Cisco CallManager Express CP3911CCME $166.74 Cisco Unified Sip Phone 3911 VoIP Phone Sip Dark Gray With 1 X User License For Cisco CallManager Express CP3911CCME

Panasonic KX-UT670 Executive SIP Phone $438.99 "Panasonic KX-UT670 Brand New Includes 2 Year Limited Warranty, The KX-UT670 is equipped with a large, 7-inch LCD touchscreen allowing easy access to features and installedapplications. The open-source operating system is programmable in Java enabling development ofthird party business applications. Users can browse the web, access email and check daily calendar appointmentsright on the phone. The UT670 can also sync with up to 16 Panasonic network security cameras so youcan monitor live security video directly from the phone. KX-UT670 Features, SIP Tablet Phone, 7"" Color Touchscreen, Android Operating System, 6 SIP Accounts, HD Video and Voice, Bluetooth Enabled, 24 Programmable Keys, 100 Entry Phonebook, 3 Way Conferencing, Programmable: JAVA, HTML & Adobe Flash, SD Card Slot, USB Port, Plug and Play Configuration, Power Over Ethernet (PoE) Support, Optional Power Adapter: KX-A422, 2 Ethernet Ports, Integration w/ IP Cameras (up to 16), Certified For Use With Following Platforms: Broadsoft Broadworks&reg, Digium Astrisk&reg"

CyberData VoIP Paging Gateway $285.99 010846 1 1 x RJ-11 FXO 1 x RJ-45 10/100Base-TX PoE 1.15" Height x 4.05" Width x 6.11" Depth 10 Mbps 10 Mbps Ethernet 10/100Base-TX 100 Mbps 100 Mbps Fast Ethernet 2 Year 48 V DC IGMP Web-based Configuration SIP RFC 3261 compatible The CyberData VoIP Paging Gateway enables access to existing paging speakers through a VoIP Phone System. The interface is designed to use an existing analog zone controller with a TAM interface and support paging up to 99 zones from a VoIP phone. CyberData CyberData Corporation G.711 Twisted Pair 10/100Base-TX VoIP Gateway VoIP Paging Gateway www.cyberdata.net

Sip Ip Dect Cordless Phone $231.08 - Panasonic IP telephone with corded base station and 1 cordless handset- VoIP support: IETF SIP version 2(RFC3261 ands companion RFCs), BroadWorks R15 / BroadSoft, Asterisk- Codec: G.711a-law / G.711m-law / G.722(wideband) / G.729a / G.726(32K)- Network

Aastra 6731i SIP VoIP Phone $110.49 "Aastra 6731i Brand New Includes One Year Warranty, Item # A6731-0131-10-01 The Aastra 6731i VOIP phone features a 3 line LCD display and supports up to 6 lines with call appearances. It offers advanced XML capability to access custom applications and is fully interoperable with leading IP-PBX platforms. The 6731i is well suited for daily telephone use in both small and large businesses. 6731i Features: 8 Programmable Line / Feature Keys, Corded Voice Over IP Phone, Voice Over Internet Protocol (VoIP), Session Initiation Protocol (SIP), Built-In HTTP Server, XML Browser, Up to 9 Call Appearance Lines, Caller ID / Call Waiting, Hearing Aid Compatible, Wall Mountable/Mount Included, Echo Cancellation, Speakerphone, 3-Line LCD Display, LED For Call &amp Message Waiting Indicator, 3-Way Conferencing, Speed Dial, Call Forwarding, Missed Call Indicator, Speakerphone Volume Control, Volume Control, Distinctive Ring Detection, Shared Call Appearances Allows You to Place a Call on Hold at One Set and Pick it up Easily at Another Set, Intercom, Call Transfer, Page, Auto Answer, Redial, Hold, Mute, 4 Navigational Keys, Do Not Disturb, Call Timer, Two-Port 10/100 Mbps Ethernet Switch, Headset Jack, Multilingual Menu Support (English, French, Spanish, Italian, &amp German), VoIP Features: Dual 10/100 Mbps Switched Ethernet Ports, Manual or DHCP IP Address Setup, Time / Date Synch Using SNTP, NAT, QOS, TOS, & Differentiated Services Code Point, Built-In HTTP/HTTPS Server For Web Administration & Maintenance, Voice Codec Support: G.711 -law / A-law, G.729, Protocols: IETF SIP (RFC3261 & Associated RFCs) The 6731i IP Phone requires the following environment: SIP-based IP PBX system or network installed and running with a SIP account created for the 6730i phone. Access to a Trivial File Transfer Protocol (TFTP), File Transfer Protocol (FTP), Hypertext Transfer Protocol (HTTP) server, or Hyper Text Transfer Protocol over Secure Sockets Layer (SSL) (HTTPS). 10/100 MB Ethernet/Fast Ethernet LAN, Category 5/5e Straight Through Cabling, Power over Ethernet (PoE) Inline Power Injector (Optional Accessory - Neces"

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