Artikel zoeken
Informatie
Nieuw
- VoIP Telephones
- ISDN Interfaces
- Headsets
- Conference Solutions
- VoIP PBX
- Gateways
- Analog Adaptor ATA
- Switches
- Navayo
- TRex
- IP Door Phones
- Fax-Server
- WLAN + DECT over SIP
- IP Cameras
- Network Accessories
- Special Offers
- With PJP-50USB firmware Version 2.000 it is now possible to chain up to four PJP-50USBs for...
- Ga verder
- An upgrade to the popular Skype-certified GXV3140, the GXV3175 features a 7” touch screen...
- Ga verder
- New Fully-Featured 2-line GXP1450 Enterprise SIP Phone Delivers Innovative Desktop...
- Ga verder
- VoIP Media Gateway voip-mg The Junghanns .NET Voice over IP Media Gateway (voip-mg) turns all...
- Ga verder
- 42com's Open Source channel driver exclusively allows the combination of the flexible Asterisk-...
- Ga verder
Artikel 13 / 18
Algemene Productinformatie
Portech MV-378-3G VoIP UMTS Gateway
ArtNr: VOPORTMV3783G
8 Ports VoIP UMTS Gateway
Dit Artikel is niet op voorraad en moet nabesteld worden.
- Description
MV-378 is a 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. It can enable to make 8 calls simultaneously from IP phones to GSM/CDMA/UMTS networks and GSM/CDMA/UMTS networks to IP phone.
Features
- VoIP(SIP),GSM conversion.(MV-372)
- VoIP(SIP),UMTS conversion.(MV-372-3G) for all world and Japan (SoftBank Mobile,Docomo)
MV-372-3G: mobile to lan 2 stage dialing-free mode.
When calling party call MV-372-3G 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 dialing
When lan phone and MV-372 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. - Support free mode-two stage dialing and assigned mode-one 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 all AT Command
- Call Back feature
- All functions can be set on web.
- 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
- Frequency:
- Quad Band:850/900/1800/1900MHZ
- 3G/UMTS Version for all world and Japen (SoftBank Mobile / Docomo)
- 3G: EDGE/GPRS 850, 900, 1800, 1900 MHz / HSDPA/UMTS 850, 1900, 2100 MHz
Artikel 13 / 18
