VoIP Accessories

The HT502 is a powerful VoIP router. The product's inclusion of an integrated high performance NAT router and dual 10/100Mbps Ethernet WAN and LAN ports enables a shared broadband connection between multiple ethernet devices. In addition to being SIP 2.0 standard compliant, the product supports Universal Plug-in-Play (UPnP), up to 2 SIP account profiles, and advanced telephony features.    Enhanced security    Automated provisioning using symmetric and asymmetric voice    Support for a broad range of popular voice codec                 Features Universal Plug-in-Play (UPnP) 2 FXS ports (RJ11) w/up to 2 SIP account profiles Dual10/100 Mpbs ports (RJ45) w/integrated router T.38 Fax HTTP/HTTPS(pending)/Telnet/TFTP Provisioning SIP over TCP/TLS IP connectivity for any phone and fax Web management for easy configuration and installation Offers traditional and advanced telephony features Portable and compact for use at home or on the road  Specifications Ethernet Ports: 2 RJ45 (LAN/WAN) NAT/Router: Yes DHCP: Client/Server FXS Port: 2 FXO Port: No PSTN Pass-through Port: No Voicemail Indicator: Yes Voice Codec: G.711(a/u-law), G.723.1, G.729A/B/E, G.726-40/32/24/16 and iLBC, T.38 fax…
HT701 is a next generation, powerful IP ATA (analog telephone adapter) for residential and road warriors. Its ultra-compact size, superb voice quality, rich functionalities, strong security protection, excellent manageability and auto provisioning, as well as unrivaled affordability enable service providers to offer high quality IP service at extremely competitive price. The HT701 is an ideal single-port ATA for large scale commercial IP voice service deployment.Single FXS telephone port(supporting 5 REN) single 10/100M Ethernet portLEDs for Power, Internet, LINK/ACT, and Phone statusStrong security protection of voice/data privacy using TLS/SRTP/HTTPS, secure and automated provisioning using TR069 and HTTP/HTTPS/TFTP             Features Single FXS telephone port(supporting 5 REN) single 10/100M Ethernet port LEDs for Power, Internet, LINK/ACT, and Phone status Strong security protection of voice/data privacy using TLS/SRTP/HTTPS, secure and automated provisioning using TR069 and HTTP/HTTPS/TFTP Advanced Telephony features including caller ID, call waiting, 3-way conference, transfer, forward, do not disturb, message waiting indication, multi-language voice prompt, T.38 Fax, flexible dial plan Support comprehensive voice codecs including G.711 with Annex I/II, G.723.1, G.729A/B, G.726,…
HT702/704 are next generation, powerful 2/4-port IP ATA (analog telephone adapter) for residential users and small businesses. Their compact size, superb voice quality, rich functionalities, strong security protection, excellent manageability and auto provisioning, as well as unrivaled affordability enable service providers to offer high quality IP voice service at extremely competitive price. The HT702 and HT704 are ideal 2/4-port ATAs for large scale commercial IP voice service deployment.   2 FXS port (HT702) or 4 FXS port (HT704), supporting 3 REN and single 10/100M Ethernet port Advanced telephony features including caller ID, call waiting, 3-way conference, transfer, forward, do not disturb, message waiting indication, multi-language voice prompt, T.38 Fax, flexible dial plan Strong security protection of voice/data privacy using TLS/SRTP/HTTPS, secure and automated provisioning using TR069 and HTTP/HTTPS/TFTP       Features 2 FXS port (HT702) or 4 FXS port (HT704), supporting 3 REN and single 10/100M Ethernet port LEDs for Power, Internet, LINK/ACT, and Phone status Advanced telephony features including caller ID, call waiting, 3-way conference, transfer, forward, do not disturb, message waiting indication,…
The GXW400x FXS Series 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 or 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                   Features 4 and 8 FXS port media gateways Two RJ-45 ports 10/100 Mbps (switched or routed) Multiple SIP accounts (choice of 2 profiles per account) T.38 Fax compliant G.168 Echo Cancellation Voice Activation Detection (VAD) Comfort Noise Generation (CNG) Packet Loss Concealment (PLC) Supports PSTN/PBX analog telephone sets or analog trunks Specifications FXS Ports: 4 FXS (GXW4004), 8FXS (GXW4008 Ethernet Ports: 2 RJ45 10/100Mbps (LAN/WAN) PSTN Line Port: 1 RJ11, for Fail-Over PSTN Voice Codecs: G.711 (a/µ law), G.723, G.726,…
The GXW FXO IP Analog Gateway series offers the small enterprise, SOHO, remote offices and multi-location enterprises a cost-effective, easy to deploy VoIP FXO solution. The GXW410x series allows any business to seamlessly connect multiple locations with up to 8 PSTN lines, to an IP PBX system, or with an existing traditional phone system.    The GXW series includes two models with 4 or 8 ports respectively    Designed and tested for full interoperability with leading IP-PBXs, soft-switches and SIP-based environments    Manageability, a simple configuration, superb voice and video quality and feature rich functionality    Based on open industry standards             Features 4 and 8 FXO port media gateways External power supply Two RJ-45 ports 10/100 Mbps (switched or routed) TFTP and HTTP firmware upgrade support Multiple SIP accounts (choice of 3 profiles per account) Programmable PSTN line settings for different countries/regions 1 or 2 stage dialing Caller ID G.168 - echo cancellation Flexible DTMF transmission: In Audio, RFC2833, SIP Info or any combination of the 3 Selectable, multiple LBR coders per channel…
The GXW4024 gateway enables small and medium businesses to create a cost-effective hybrid IP and analog telephone systems and enjoy the benefits of VoIP communications while preserving investment on existing analog phones and traditional PBX systems.    High density SIP based analog telephone VoIP gateway    Fully interoperable with leading IP-PBX and Softswitch systems    Features 24 telephone ports both RJ11 and 50-pin Telco connector                     Features 24 telephone FXS ports with both RJ11 and 50-pin Telco connector Up to 2 SIP server profiles per system and independent account per port Supported voice codecs include G.711, G.723, G.726A/B/E, iLBC, T.38 Fax Carrier grade G.168 line echo cancellation Ideal IP enabler for analog phones, faxes and legacy PBX systems Specifications Telephone Interface: 24 x RJ11 FXS ports and 1 x 50-pin Telco connector Network Interface: 1 x 10M/100 Mbps auto-sensing RJ45 port Audio Codecs: G.711, G. 723, G.726 (40/32/24/16), G.729A/B/E, iLBC Security: SRTP, TLS/SIPS Fax over IP: T.38 compliant Group 3 Fax Relay up to 14.4 kbps and auto switch…
  Phone Adapter with 2 FXS Ports for Voice-over-IP (Europe)  The incredible cost savings of Internet telephone service isn't limited only to those with IP phones. Part of Cisco Small Business Voice Gateways and ATAs, the PAP2T offers the benefits of high-quality voice over IP (VoIP) without the need to upgrade your existing analog phones.The PAP2T features two standard analog, fax-friendly telephone ports which operate as separate lines, complete with individual phone numbers. All the features you've come to expect are here, such as call waiting, caller ID, voicemail, and more. It's easy to select your preferred free local-dialing area code, or even add a virtual number from anywhere to be forwarded to your Internet phone.         Additional features of the Cisco PAP2T Internet Phone Adapter include:    * High-quality, feature-rich telephone service over your broadband Internet connection    * Two standard telephone ports for analog phones or fax machines that serve as separate lines, complete with unique phone numbers    * Up to 256-bit AES encryption for data transmissions and password-protected administrative options    *…
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…
  Phone Adapter with 1 FXS & 1 FXO Ports Analog Calls Routed Over the InternetInternet telephone service is now more accessible than before. Part of Cisco Small Business Voice Gateways and ATAs, the SPA3102 Voice Gateway allows automatic routing of local calls from mobile phones and land lines to Voice over Internet Protocol (VoIP) service providers, and vice versa. By first calling the local number on the SPA3102, you can reduce or eliminate long-distance charges altogether.If power goes out or Internet service is lost, calls can be safely redirected to a traditional carrier via the standard analog interface.         Additional features of the Cisco SPA3102 include:    * One RJ-11 POTS (Plain Old Telephone Service) FXS port to connect an existing analog phone or fax machine    * One PSTN FXO port to connect to a Telco or PBX circuit    * Two 100BaseT RJ-45 Ethernet interfaces to connect to a home or office LAN, and an Ethernet connection to a broadband modem or router    * FXS and FXO lines that can be independently…
  Phone Adapter with 8 FXS Ports High-Density Voice SolutionDon't let analog phones come between you and Internet telephony. Part of Cisco Small Business Voice Gateways and ATAs, the SPA8000 8-Port IP Telephony Gateway allows for connections between eight analog telephones and an IP-based data network, providing small businesses with enhanced communication services through a broadband Internet connection.Ideal for multi-dwelling environments or business settings such as call centers, the SPA8000 converts voice traffic and faxes into data packets for IP transmission.         Additional features of the Cisco SPA8000 include:    * Eight RJ-11 FXS ports to connect analog telephones to IP-based data networks    * A single multiport RJ-21 50-pin connector, offering an alternative connection choice    * One 10/100 Base-T RJ-45 Ethernet interface to connect to either a router or multilayer switch    * Toll-quality voice and carrier-grade feature support    * Large-scale deployment management    * Strong security with reliable, encryption-based methods for communication, provisioning, and servicing   more here
IP Telephony Gateway με 4 FXS και 4 FXO Ports The Cisco SPA8800 IP Telephony Gateway is a multipurpose solution that small businesses can use to connect an on-premise IP private branch exchange (PBX) system to the public switched telephone network (PSTN), or connect a legacy time-division multiplexing (TDM) PBX or key system to voice-over-IP (VoIP) services. The Cisco SPA8800 provides a combination of four Foreign eXchange Office (FXO) and four Foreign eXchange Subscriber (FXS) ports to connect to existing analog phones, which helps protect and extend your existing communications equipment investments, too.The Cisco SPA8800 also has the capability to be configured as an FXO gateway for an Asterisk Open Source PBX providing a versatile solution when conditions favor an external device.       Additional features of the Cisco SPA8800 include:Converts voice traffic into data packets for transmission over an IP networkSession Initiation Protocol (SIP) standards for voice and data networking provides reliable voice and fax operationSecure, encryption-based methods for communicating, provisioning, and servicingToll-quality voice and carrier-grade feature supportLarge-scale deployment management   more here

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…

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…

2N Helios Easy Gate

Written by Friday, 23 November 2012 10:37
2N® VoiceBlue NextTeach Your IP Exchange to Save MoneyThe 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…

ITS CGW GSM

Written by Friday, 23 November 2012 10:41
The CGW- VIP is an advanced Cellular-to-VoIP gateway. Unique by its pure-IP and fully managed platform specifically designed to handle both voice and managed data traffic, with the ability to ensure priority and quality of service for voice calls.The CGW-VIP has several connectivity options and many useful applications. It suits perfectly to offices with an IP-PBX as a cost saving Cellular-to-VoIP and VoIP-to-Cellular Gateway. And it can be used to establish a virtual private voice network between branches with toll-free calls.More Of CGW-VIP ApplicationsBusiness continuityLast mile access tool for underserved areasA virtual private voice network for branchesA costs saving gateway       Main FeaturesSupports VoIP, SIP, and GSM connectionsInteroperates with all standard IP-PBX/soft switch equipmentEmbedded web serverLocal and remote management via user-friendly web interfaceLocal and remote upgrades of configuration files and firmwareSupports registration of SIP extensionsEnables creation of a virtual office with remote extensionsBuilt-in firewall and NATBuilt-in Least-Cost-Routing (LCR)CLIP/CLIRInterfacesLAN / WAN 10/100 Base-T, RJ-45VoIPVoice signaling: SIPLocal SIP server/registerCodecs: G.711 PCM (μ/A-Law), G.729*Echo canceler: G.168-2002CellularGSM module: integrated quad band 850/900/1800/1900 MHzGSM module: 2Antenna: 2 external…

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