VoIP GSM шлюз Portech MV-374 на 4 SIM карты (447330001) - Аукцион Online

Внимание! В связи с естественной периодичностью поставок, поставка данного лота конечному покупателю может занять до 14 рабочих дней. Извините за неудобства!

За подробными консультациями по данному лоту обращайтесь по телефону 80679586868.

MV-374 is a 4 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,SIP Proxy Server,VoipBuster. It can enable to make 4 calls simultaneously from IP phones to GSM/CDMA/UMTS networks andGSM/CDMA/UMTS network to IP phone.

Возможности

1. VoIP(SIP)、GSM conversion.(MV-374)

2.
VoIP(SIP)、CDMA conversion.(MV-374C) - CDMA 2000(800MHz)

3.
VoIP(SIP)、UMTS conversion.(MV-374U) for all world and Japan (SoftBank Mobile)
MV-374U: mobile to lan 2 stage dialing-free mode.
When calling party call MV-374U 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**

4.
50 sets of LAN --> MOBILE routes setting，50 sets of MOBILE --> LAN routes setting.
-Support one stage diaing
*When lan phone and MV-374 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 free mode-two stage dialing and assigned mode-one stage dialing

5. Voice response for setting and status(dial in from mobile).

6. For call termination (VoIP to GSM/CDMA/UMTS ) and origination ( GSM/CDMA/UMTS to VoIP).

7. Standard SIP(RFC2543，RFC3261) protocol，communicates with other gateway or PC

8. Receive SMS and Send SMS (CDMA version,sms feature is unavailable)

9. Allows your program Send/receive SMS with AT Command

10. Call Back feature

11. All functions can be set on web.

Спецификации

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.723.1 (5.3k)、G.723.1 (6.3k)、G.729A、G.729A/B
Voice Quality、VAD、CNG、AEC、LEC、Packet loss

Frequecny：
Dual BAND:900/1800 MHZ
Tri Band: 900/1800/1900MHZ
Tri Band: 850/1800/1900MHZ
Quad Band:900/1800/1900/850MHZ
3G/UMTS: for all world and Japan (SoftBank Mobile)
3G:EDGE/GPRS 850, 900, 1800, 1900 MHz / HSDPA/UMTS 850, 1900, 2100 MHz
CDMA 2000(800MHZ)

	Увеличить	VoIP GSM шлюз Portech MV-374 на 4 SIM карты (номер 447330001)
	Увеличить	Как покупать? Описание

Оплата и доставка лота
Способы доставки		Способы оплаты и стоимость доставки
Заказная отправка почтой Курьерская служба Передача лично		Полная предоплата - стоимость доставки : нет данных Другое (смотрите Подробнее о доставке и оплате лота)
Подробнее о доставке и оплате лота
Возможна оплата безналичным расчетом.

1.	VoIP(SIP)、GSM conversion.(MV-374)
2.	VoIP(SIP)、CDMA conversion.(MV-374C) - CDMA 2000(800MHz)
3.	VoIP(SIP)、UMTS conversion.(MV-374U) for all world and Japan (SoftBank Mobile) MV-374U: mobile to lan 2 stage dialing-free mode. When calling party call MV-374U 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**
4.	50 sets of LAN --> MOBILE routes setting，50 sets of MOBILE --> LAN routes setting. -Support one stage diaing When lan phone and MV-374 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 free mode-two stage dialing and assigned mode-one stage dialing*
5.	Voice response for setting and status(dial in from mobile).
6.	For call termination (VoIP to GSM/CDMA/UMTS ) and origination ( GSM/CDMA/UMTS to VoIP).
7.	Standard SIP(RFC2543，RFC3261) protocol，communicates with other gateway or PC
8.	Receive SMS and Send SMS (CDMA version,sms feature is unavailable)
9.	Allows your program Send/receive SMS with AT Command
10.	Call Back feature
11.	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.723.1 (5.3k)、G.723.1 (6.3k)、G.729A、G.729A/B Voice Quality、VAD、CNG、AEC、LEC、Packet loss
	Frequecny： Dual BAND:900/1800 MHZ Tri Band: 900/1800/1900MHZ Tri Band: 850/1800/1900MHZ Quad Band:900/1800/1900/850MHZ 3G/UMTS: for all world and Japan (SoftBank Mobile) 3G:EDGE/GPRS 850, 900, 1800, 1900 MHz / HSDPA/UMTS 850, 1900, 2100 MHz CDMA 2000(800MHZ)