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 have credit. -Support assigned mode and free mode-two stage dialing Voice response for setting and status(dial in from mobile). For call termination (VoIP to GSM/CDMA/UMTS) and origination (GSM/CDMA/UMTS to VoIP). Standard SIP(RFC2543,RFC3261) protocol,Communicates with other gateway or PC Receive SMS and Send SMS (CDMA version,sms feature is unavailable) Allows your program Send/receive SMS with AT Command Call Back feature All functions can be set on web. Provide CDR 24 months warranty Specification Protocols:SIP (RFC2543,RFC3261) TCP/IP:IP/TCP/UDP/RTP/RTCP/CMP/ARP/RARP/SNTP/DHCP/DNS Client,IEEE802.1P/Q,ToS/DiffServ,NAT Traversal,STUN,uPnP,IP Assignment,Static IP,DHCP,PPPoE Codec:G.711 u-Law,G.711 a-Law,G.729A,G.729A/B Voice Quality,VAD,CNG,AEC,LEC,Packet loss Frequecny: Quad Band:850/900/1800/1900MHZ 3G/UMTS: for all world and Japan (SoftBank Mobile,Docomo) 3G/UMTS Version: 3G:EDGE/GPRS 850, 900, 1800, 1900 MHz / HSDPA/UMTS 850, 1900,2100 MHz CDMA 2000(800MHZ/1900MHZ) **Please note** 1. Most CDMA -2000 operators don't offer Answer signal. So VoIP to Mobile, MV-370 will connect soon. CDMA -2000 operators will start billing soon. It doesn't wait mobile side answer 2. CDMA Version doesn't support SMS Feature and 180/183 unavailable 3. CDMA version doesn't have remote SIM feature 4. CDMA version doesn't support DTMF
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 design and compatible with international voice and data standards, the Cisco SPA122 can be used with residential, home-office, and small business VoIP service offerings, including full-featured hosted or open source IP PBX environments. This easy-to-use solution delivers advanced features to better connect employees and serve customers, all on a highly secure Cisco network.The Cisco SPA122 ATA with Router (Figures 1 and 2):• Enables high-quality VoIP service with a comprehensive feature set through a broadband Internet connection• Provides high-quality, clear-sounding voice, using advanced voice quality-of-service (QoS) capabilities and the industry-leading voice Session Initiation Protocol (SIP) stack• Supports reliable faxing with simultaneous voice and data use• Includes two standard telephone ports, each with an independent phone number, for use with fax machines or analog phone devices, and one fast Ethernet WAN port, and one fast Ethernet LAN port for local home or business network connection• Is compatible with all industry voice and data standards and common telephone features such as caller ID, call waiting, and voicemail• Includes a simple-to-use web-based configuration utility for easy deployment
The IP convergence system HiPath 3000 offers medium sized enterprises of 10 to 500 subscribers high reliability voice communications using high quality terminals with easy operation.HiPath 3000 V9 can be enhanced with the server based Unified Communications Solution OpenScape Office This provides an easy way from voice to UCC .
New Features for HiPath 3000 V9
OpenScape Office HX to connect to MX/LX networks Enhanced features of OpenScape Office V3 Network-wide unified communications (presence, call status, instant messaging, directory...) SIP attack protection Enhanced remote administration: SSDP support via Shiva Plug optiClient Attendant call park improvements HiPath Cordless user enhancements incl. MWI indication
Capacity:
Up to 250 channels (ISDN, loop start, IP) Up to 384 digital subscribers Up to 500 IP workpoints HiPath Cordless with up to 64 base stations and 250 handsets