Extremely light-weight design, one of the lightest in the world. Stay put click stop headband for a perfect snug fit. Made of high strength engineering plastics. Highest degree of flexibility and float to ensure the best fit on all types, shapes and sizes of head. Best in class flexibility of boom mic tube ensuring perfect positioning of the boom mic on all call conditions. Compatible with all types of phones systems such as digital, IP and analog.

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- Adjustable Surgical Steel Headband - Flexible geared Bi-Metal Mic Boom - Noise cancelling Microphone - Soft leatherette ear cushions Learn More

IP PBX : 200 subscribers

*ast e200 is the *asterisk based IPPBX solution which support 200 Subscribers. *astTECS IPPBX which is the new generation technology in PBX world and also used widely in developed countries, IPPBX to help you easily transition from simple telephony to feature-rich UC solution. Connect with various interfaces. *astTECS IPPBX will connect all your remote offices through VOIP.
FREE REMOTE INSTALLATION from *astTECS FREE REMOTE SUPPORT - 24X7 (For 3 Months) from *astTECS

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* IPPBX 200 Extension. * Asterisk Installed/Open Source. * Easy Setup & Management. * No licensing Fees * Connect with various interfaces like PRI, Analog, GSM, VOIP. * One year Hardware AMC. Download Brochure Learn More

astTECS PRI Gateway is a compact, dedicated and feature-rich VoIP to T1/E1 PRI gateway. It is a dual-span gateway offering 60 simultaneous calls.It is a cost-effective trunk gateway equipment specifically designed for open source market which is ideal for open source based solutions such as Asterisk/Elastix/Trixbox.

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Product Highlights

• 2 dual-color line keys (with 2 SIP accounts and up to 2 call appearances), 3 XML programmable context-sensitive soft keys, 3-way conferencing, multi-language support
• Automated personal information service (e.g., local weather, etc.), personalized music ring tone/ring back tone, customizable screen content/format using XML, advanced web and enterprise applications integration (pending)
• Dual-switched 10/100 Mbps ports, integrated PoE on GXP1625
• Use with Grandstream’s UCM6100 series IP PBX appliance for Zero-Config provisioning, 1-touch call recording and more
• HD wideband audio, superb dull-duplex hands-free speaker- phone with advanced acoustic echo cancellation and excellent double-talk performance
• Automated provisioning using TR-069 or encrypted XML configuration file, SRTP and TLS for advanced security protection, 802.1x for media access control
• 132 x 48 pixel backlit graphical LCD display
• Large phonebook (up to 500 contacts) and call history (up to 200 records)
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Product Highlights

• 2 dual-color line keys (with 2 SIP accounts and up to 2 call appearances), 3 XML programmable context-sensitive soft keys, 3-way conferencing, multi-language support
• Automated personal information service (e.g., local weather, etc.), personalized music ring tone/ring back tone, customizable screen content/format using XML, advanced web and enterprise applications integration (pending)
• Dual-switched 10/100 Mbps ports, integrated PoE on GXP1625
• Use with Grandstream’s UCM6100 series IP PBX appliance for Zero-Config provisioning, 1-touch call recording and more
• HD wideband audio, superb dull-duplex hands-free speaker- phone with advanced acoustic echo cancellation and excellent double-talk performance
• Automated provisioning using TR-069 or encrypted XML configuration file, SRTP and TLS for advanced security protection, 802.1x for media access control
• 132 x 48 pixel backlit graphical LCD display
• Large phonebook (up to 500 contacts) and call history (up to 200 records)
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Product Highlights

• Single SIP account, up to 2 call appearances, 3 XML programmable context-sensitive soft keys, 3-way conferencing, multi-language support
• Personalized music ring tone/ring back tone and integration with advanced Web and enterprise applications, local weather service
• Dual-switched 10/100 Mbps ports
• Use with Grandstream’s UCM6100 series IP PBX appliance for Zero-Config provisioning, 1-touch call recording and more
• Full-duplex hands-free speakerphone with advanced acoustic echo cancellation, Electronic Hook Switch (EHS) with Plantronics headsets
• Automated provisioning using TR-069 or encrypted XML configuration file, SRTP and TLS for advanced security protection, 802.1x for media access control
• 132 x 48 pixel LCD display
• Large phonebook (up to 500 contacts) and call history (up to 200 records)
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The GXW4216 is a next generation high performance high-density analog VoIP gateway that is fully compliant with SIP standard and interoperable with various VoIP systems, analog PBX and phones on the market. The GXW42XX series gateway offers small and medium businesses a cost-effective hybrid IP and analog telephone system that allows them to enjoy the benefits of VoIP communications while preserving investment on existing analog phones, Fax machines and legacy PBX systems.

Product Features: *Multiple FXS analog telephone ports *Superb voice quality *Rich telephony functionalities, easy provisioning *Flexible dialing plans, advanced security protection *Strong performance in handling high volume voice calls. Learn More

Grandstream 8 Port FXS Gateway is an ideal solution for businesses looking to connect one or more lines of a traditional PBX to a VOIP phone system or provider. The GXW400x features 8-port FXS interfaces for analog telephones, dual 10M/100M network ports with integrated router, PSTN life line in case of power failure, and an RS232 serial port for administration.

- Support for 2 SIP account profiles, caller ID for various countries/regions - Support for T.38 fax, flexible dialing plans, security protection (SIPS/TLS), and comprehensive voice codecs Learn More

Grandstream 4 Port FXS Gateway is an ideal solution for businesses looking to connect one or more lines of a traditional PBX to a VOIP phone system or provider. The GXW400x features 4-port FXS interfaces for analog telephones, dual 10M/100M network ports with integrated router, PSTN life line in case of power failure, and an RS232 serial port for administration.

- Support for 2 SIP account profiles, caller ID for various countries/regions - Support for T.38 fax, flexible dialing plans, security protection (SIPS/TLS), and comprehensive voice codecs Learn More

Caudalfin Single Span E1/T/J1 cards are high performance, cost effective telephony cards available with carrier grade optional hardware echo cancellation. The Caudalfin Single Span E1/T/J1 cards allows up to 24 (T1) or 30 (E1) channels and allows integration of legacy telephony systems with emerging Voice-over IP (VoIP) technologies.

Caudalfin Single Span E1/T/J1 cards are designed to work with Open Source based VoIP platforms such as Asterisk® (e.g. PBX in a Flash®, FreePBX® Elastix®) & FreeSWITCH®. It supports industry standard telephony protocol families. Both line-side and trunk-side interfaces along with advanced call features.

Technical Specifications:

Line Protocols

Dimensions:	2U rack/server mountable Tapping capable Ease of installation (no patches required)
Operating Systems	Linux Kernel: 2.4 to Latest, Windows: XP/Vista/Win7
Voice CAS, MRC/R2 PRI, SS7,HDLC / PPP
Framing Types	T1 / E1 / J1 with D4 / ESF Signaling – CAS / CCS  CRC4 / Non-CRC4
Minimum Hardware Requirement	1.6-GHz Pentium IV, 512 MB RAM,3.3V or 5V PCI-e x1 slot

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