VoIP Accessories

Designed for users looking to connect their analog devices to a VoIP network at home or in the office. The HT802 is a powerful analog telephone adapter that is easily deployable and manageable. It comes equipped with 2 FXS ports to create a high-quality network solution. Description The HT802 delivers powerful VoIP technology and routing capabilities to home and office environments, and allows users to successfully connect their analog devices to a manageable and powerful VoIP network. Built upon Grandstream’s market-leading SIP ATA/gateway technology, with millions of units successfully deployed worldwide, this powerful ATA features exceptional voice quality in various applications and environments. The HT802 comes with 2 easy-to-use FXS ports, state-of-the-art encryption with a unique security certificate per unit, automated provisioning for volume deployment and device management and outstanding network performance. Features     Supports 2 SIP profiles through 2 FXS ports and a single 10/100Mbps port    TLS and SRTP security encryption technology to protect calls and accounts    Automated provisioning options include TR-069 and XML config files    Supports 3-way voice conferencing    Failover SIP server automatically…
Designed for users looking to connect their analog devices to a VoIP network at home or in the office. The HT812 is a powerful analog telephone adapter that is easily deployable and manageable. It comes equipped with 2 FXS ports and an integrated Gigabit NAT router. Description The HT812 delivers powerful VoIP technology and routing capabilities to home and office environments, and allows users to successfully connect their analog devices to a manageable and powerful VoIP network. Built using Grandstream’s market-leading SIP ATA/gateway technology, with millions of units successfully deployed worldwide, this powerful ATA features exceptional voice quality in various applications and environments. The HT812 comes with 2 easy-to-use FXS ports, an integrated Gigabit NAT router, state-of-the-art encryption with a unique security certificate per unit, automated provisioning for volume deployment and device management and outstanding network performance. Features     Supports 2 SIP profiles through 2 FXS ports and dual Gigabit ports    Includes a built-in NAT router which can handle routing speeds up to 100MBps    TLS and SRTP security encryption technology to protect calls and accounts   …
The HT813 is an analog telephone adapter that features 1 analog telephone FXS port and 1 PSTN line FXO port. Description The HT813 is an analog telephone adapter that features 1 analog telephone FXS port and 1 PSTN line FXO port in order to offer backup lifeline support using a PSTN line. The integration of a FXO and FXS port enables this hybrid ATA to support remote calling to and from the PSTN line. For added flexibility, the FXS port extends VoIP service to one analog device. Users can convert their analog technology to VoIP thanks to the HT813’s ultra-compact size, HD voice quality, advanced VoIP functionality, high-end security protection and multiple auto provisioning options. These advanced features also allow service providers to offer high quality IP service to customers looking to upgrade to VoIP Features     Supports 2 SIP profiles through 1 FXS port and 1 FXO port    Dual 100Mbps LAN and WAN ports    Lifeline support (FXS port will be hard-relayed to FXO port) in case of power outage    3-way voice conferencing per port   …
Designed for home or office users looking to connect their analog devices to a VoIP network, the HT814 is a powerful analog telephone adapter that is easily deployable and manageable. It comes equipped with 4 FXS ports and an integrated Gigabit NAT router. Description The HT814 delivers powerful VoIP technology and routing capabilities to home and office environments, and allows users to successfully connect their analog devices to a manageable, robustnetwork. Built using Grandstream’s market-leading SIP ATA/gateway technology, with millions of units successfully deployed worldwide, this powerful ATA features exceptional voice quality in various applications and environments. The HT814 comes with 4 easy-to-use FXS ports, an integrated Gigabit NAT router, state-of-the-art encryption with a unique security certificate per unit, automated provisioning for volume deployment and device management and outstanding network performance. Features     Supports 2 SIP profiles through 4 FXS ports and dual Gigabit ports    Includes a built-in NAT router which can handle routing speeds up to 100MBps    TLS and SRTP security encryption technology to protect calls and accounts    Automated provisioning options include TR-069 and…
Built for users looking a strong analog-to-VoIP converter, the HT818 is a powerful 8-port VoIP gateway with 8 FXS ports and an integrated Gigabit NAT router. Description The HT818 is a powerful 8-port VoIP gateway with 8 FXS ports and an integrated Gigabit NAT router. Built for users looking for a strong analog-to-VoIP converter, it features Grandstream’s market-leading SIP ATA/gateway technology with millions of units successfully deployed worldwide. This powerful gateway carries exceptional voice quality in various application environments, strong encryption with unique security certificate per unit, automated provisioning for volume deployment and device management, and outstanding network performance for enterprise use. Features     Supports 2 SIP profiles and 8 FXS ports    Strong AES encryption with security certificate per unit    Automated & secure provisioning options using TR069    3-way voice conferencing per port    Exceptional voice quality with wide-band HD codec    Supports T.38 Fax for reliable Fax-over-IP    Supports dual Gigabit network ports    High performance NAT router more here
Affordable and Feature-Rich Voice over IP (VoIP)HighlightsEliminate compromise on voice quality or features for phone and fax capabilities associated with Internet voice over IP (VoIP) service. Cisco® VoIP solutions provide the quality, peace of mind, and investment protection at an affordable price.                   Product OverviewThe Cisco SPA122 ATA with Router combines VoIP services with an internal router for LAN connectivity. Easy to install and use, it works over an IP network to connect analog phones and fax machines to a VoIP service provider and provides support for additional LAN connections.The Cisco SPA122 includes two standard telephone ports to connect existing analog phones or fax machines to a VoIP service provider. It also includes two 100BASE-T RJ-45 Ethernet ports for WAN and LAN connectivity. Each phone line can be configured independently. With the Cisco SPA122, users can protect and extend their investment in their existing analog telephones, conference speakerphones, and fax machines, as well as control their migration to IP voice with an extremely affordable, reliable solution.Compact in…

MV 370 - 372

Written by Friday, 23 November 2012 10:28
1 or 2 Ports VoIP GSM Gateway / VoIP CDMA Gateway/VoIP UMTS GatewayMV-370 -372  is a 1 Port VoIP GSM/CDMA/UMTS Gateway for call termination (VoIP to GSM/CDMA/UMTS ) and origination (GSM/CDMA/UMTS to VoIP). Support Asterisk,Trixbox,3CX,SIP Proxy Server,VoipBuster. Major Function     VoIP(SIP),GSM conversion.(MV-370)    VoIP(SIP),CDMA conversion.(MV-370C) - CDMA 2000(800/1900MHz)    VoIP(SIP),UMTS conversion.(MV-370U) for all world and Japan (SoftBank Mobile,Docomo)    MV-370U: mobile to lan 2 stage dialing-free mode. When calling party call MV-370U sim card,the calling party will hear dial tone and enter any destination number.    **How to differentiate mobile to lan-2 stage dialing is available?**    UMTS Mobile call UMTS Mobile: when the called party answer, the calling party press any DTMF.    If the called party hear DTMF Voice,this feature is available;contrariwise**    50 sets of LAN --> MOBILE routes setting,50 sets of MOBILE --> LAN routes setting.    -Support one stage diaing    *When lan phone and MV-370 both register SIP proxy Server or Asterisk or VoipBuster,you can dial any destination number from lan phone directly.    *Please note,SIP proxy Server,Asterisk need to have the route of destination number. VoipBuster need to…

MV 374 - 378

Written by Friday, 23 November 2012 10:31
4 or 8 Channels VoIP GSM/CDMA/UMTS GatewayMV-374 - 378 is a 4 or 8 channels VoIP GSM/CDMA/UMTS Gateway for call termination (VoIP to GSM/CDMA/UMTS ) and origination ( GSM/CDMA/UMTS to VoIP). It is SIP based and compatible with Asterisk,Trixbox,3CX,SIP Proxy Server,VoipBuster.Option SBK-32 :32 SIMs Remote SIM Bank and SIM Server Connect with PORTech GSM Gateway via internet You can deploy your GSM Gateway in different locations. Centralize and supervise all SIMs in one place.  Work with Dial peer Server (free)1.Dial Peer Server can manage 128 GSM ports at same time   User just need to set one SIP trunk   Send call to dial peer IP:5060 port from Asterisk/IP PBX   The call automatically switches from a busy line to available line.2.Provide CDR3.Monitor the signal of all GSM ports Major Function1.VoIP(SIP),GSM conversion.(MV-378)2.VoIP(SIP),CDMA conversion.(MV-378C) - CDMA 2000(800/1900MHz)VoIP(SIP),UMTS conversion.(MV-378U) for all world and Japan (SoftBank Mobile/Docomo)MV-378U: mobile to lan 2 stage dialing-free mode.3.When calling party call MV-378U sim card,the calling party will hear dial tone and enter any destination number.**How to differentiate mobile to lan-2 stage dialing is available?**UMTS Mobile…

2N Helios Easy Gate

Written by Friday, 23 November 2012 10:37
2N® VoiceBlue NextTeach Your IP Exchange to Save Money The 2N® VoiceBlue Next, a successor to the 2N VoiceBlue Lite gateway, belongs to the new generation of VoIP gateways. It brings you a significant reduction in the costs of calls to mobile networks and is compatible with a wide range of IP PBXs. Its simple and intuitive setting via a web interface enables you to quickly set up the VoIP interface as well as the gateway itself. The 2N® VoiceBlue is a GSM gateway designed with the goal to reduce costs as much as possible. This is ensured by an automated saving solution able to route the call according to time, pre-code or free minutes. The CallBack service is able to reduce roaming fees on trips abroad. If your company has other branch offices in several countries, you will greatly appreciate the possibility to call local mobile networks completely free of charge using VPN tariffs.Compared to our competitors’ products, the 2N® VoiceBlue offers you a unique advantage – the 2N® Mobility Extension feature - by…

ITS Door Phones

Written by Friday, 23 November 2012 10:45
Metal Single ButtonPiezo Single ButtonPiezo Keypad Aluminum case Metal single button Optional B&W or Color Video Camera Outdoor/Indoor Installation Illuminated nameplate Speed dial button more here

2N Helios Door Phones

Written by Friday, 23 November 2012 10:53
Telephone based door entry solution2N Helios is a door entry system which meets the highest demands and is of a modern design. This user-friendly door phone replaces door systems and offers many additional features. 2N Helios can be connected to any analog PBX extension available on the market. As a result of its modularity it enables the creation of customised solutions according to the user´s needs, not only in terms of its size, but also with various control elements and functions. Each basic module can be equipped with a camera, display or card reader and an anti-vandal solution against damage. 2N Helios is made of high-quality stainless steel, and its flat design makes possible installation without cutting into the wall. The main advantages of 2N Helios door entry solution include easy installation, state of art functions and a modern design.      * Unique design* Communication over telecom base lines* Calling from entrance door to your office telephones* Door opening from standard or mobile telephone,by key code or proximity card* Modular system – possibility of…

Slican Door Phones

Written by Friday, 04 October 2013 11:13
Characteristic of Slican DPH doorphones Slican DPH doorphones connected to PBX enable to establish call between person near doorphone and PBX subscriber, as well as enter to protected area. It is also possible to trigger an electrolock, for opening door or gate, from phone connected to PBX. Following functions are available in DPH doorphones:                       establish phone call using touch button two relays trigger electrolock (EZ) additional relay (STA) power supply with 12-25VAC or 14-35VDC temperature range -20 to +70 protection level - according PN-EN 60529, IP34 for DPH and IP20 for power supply. inputs COD (opening door sensor) i POD (opening door button). Both buttons operate with electrolock. EZ trigger (opening) by receiving "*" character in DTMF code embedded RFID card reader (not all models) possibility to trigger EZ by registered RFID card approach STA trigger (second relay) by receiving "1" character in DTMF code. Single pulse with fixed duration time is generated Doorphone parameters programming from phone using DTMF code more here

Grandstream GXW 4216

Written by Thursday, 07 January 2021 20:13
The GXW4200 high-density FXS gateway series enables businesses of all sizes to create an easy-to-deploy VoIP solution that takes advantage of Gigabit speeds. These FXS gateways offer the ability to seamlessly connect multiple locations and all devices within an office to any hosted or on premise IP PBX network to make deployments as easy as possible. Description Deploy the GXW4200 series to allow any businesses to create 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. The GXW4216/24/32/48 are fully compliant with SIP standard and interoperable with various VoIP systems, analog PBX and phones on the market. It features multiple FXS analog telephone ports, superb voice quality, a suite of telephony functionalities, easy provisioning, flexible dialing plans, advanced security protection and strong performance in handling high volume voice calls. The GXW42XX series gateways offers businesses a cost-effective hybrid IP and analog telephone system that allows them to enjoy the benefits of VoIP communications.…

Grandstream GXW 4224

Written by Thursday, 07 January 2021 20:26
The GXW4200 high-density FXS gateway series enables businesses of all sizes to create an easy-to-deploy VoIP solution that takes advantage of Gigabit speeds. These FXS gateways offer the ability to seamlessly connect multiple locations and all devices within an office to any hosted or on premise IP PBX network to make deployments as easy as possible. Description Deploy the GXW4200 series to allow any businesses to create 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. The GXW4216/24/32/48 are fully compliant with SIP standard and interoperable with various VoIP systems, analog PBX and phones on the market. It features multiple FXS analog telephone ports, superb voice quality, a suite of telephony functionalities, easy provisioning, flexible dialing plans, advanced security protection and strong performance in handling high volume voice calls. The GXW42XX series gateways offers businesses a cost-effective hybrid IP and analog telephone system that allows them to enjoy the benefits of VoIP communications.…

Grandstream GXW 4232

Written by Thursday, 07 January 2021 20:27
The GXW4200 high-density FXS gateway series enables businesses of all sizes to create an easy-to-deploy VoIP solution that takes advantage of Gigabit speeds. These FXS gateways offer the ability to seamlessly connect multiple locations and all devices within an office to any hosted or on premise IP PBX network to make deployments as easy as possible. Description Deploy the GXW4200 series to allow any businesses to create 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. The GXW4216/24/32/48 are fully compliant with SIP standard and interoperable with various VoIP systems, analog PBX and phones on the market. It features multiple FXS analog telephone ports, superb voice quality, a suite of telephony functionalities, easy provisioning, flexible dialing plans, advanced security protection and strong performance in handling high volume voice calls. The GXW42XX series gateways offers businesses a cost-effective hybrid IP and analog telephone system that allows them to enjoy the benefits of VoIP communications.…

Grandstream GXW 4248

Written by Thursday, 07 January 2021 20:28
The GXW4200 high-density FXS gateway series enables businesses of all sizes to create an easy-to-deploy VoIP solution that takes advantage of Gigabit speeds. These FXS gateways offer the ability to seamlessly connect multiple locations and all devices within an office to any hosted or on premise IP PBX network to make deployments as easy as possible. Description Deploy the GXW4200 series to allow any businesses to create 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. The GXW4216/24/32/48 are fully compliant with SIP standard and interoperable with various VoIP systems, analog PBX and phones on the market. It features multiple FXS analog telephone ports, superb voice quality, a suite of telephony functionalities, easy provisioning, flexible dialing plans, advanced security protection and strong performance in handling high volume voice calls. The GXW42XX series gateways offers businesses a cost-effective hybrid IP and analog telephone system that allows them to enjoy the benefits of VoIP communications.…

Grandstream GXW 4500

Written by Thursday, 07 January 2021 21:04
The GXW4500 series of Digital VoIP Gateways offers three models that provide 1, 2 or 4 software configurable E1/T1/J1 spans and support 30, 60, or 120 concurrent calls. Description The GXW4500 series are E1/T1 Digital VoIP Gateways that allow digital PSTN and ISDN trunks to be integrated with VoIP networks. By connecting the GXW4500 series with a VoIP network and a traditional PBX or E1/T1/J1 provider, businesses can drastically increase the amount of PSTN/ISDN trunks integrated with their VoIP network. The GXW4500 series offers three models that provide 1, 2 or 4 T1/E1/J1 spans and support 30, 60 or 120 concurrent calls to cater to the VoIP needs of large and medium sized enterprises. The GXW4500 series supports PRI, SS7 and MFC R2 digital signaling protocols and nearly all major voice codecs including Opus, G.722, G.711, G.729, G.723, iLBC, G.726, GSM-FR, etc. This digital VoIP gateway series includes dual Gigabit network ports with a configurable integrated NAT router, two USB 3.0 ports, and one SD card interface. By adding echo cancellation, jitter buffer, T.38 fax…