PPPoE VoIP GSM Gateway SMS for WAN Connection , Voip Call Termination Specification: # Brand name: DBL # Model Number: GoIP 1 # Type: VoIP GSM Gateway # SIM Cards : 1 sim card # Size: 13*10*6 cm # Protocol: SIP & H.323 # Support: SMS Server , VPN Quick detail: 1. Quad band 2. Bulk SMS/USSD 3. IMEI changeable 4. Support SIM Bank 5. VoIP SIP&H323 6. Remote Access 7. Relay and VPN 8. ASR &ACD monitoring 9. Optional SMS termination 10.Auto Balance and Recharge The GoIP series gateway is a broadband relay gateway newly developed by DBL Technology. It is a new product for seamless connection between the GSM network and VoIP network. When the mobile phone SIM card is installed in the GoIP, users can register the GSM phone to the VoIP softswitch system. Through the GoIP, users can realize the uplink and downlink calls between the GSM network and the VoIP network. In addition, the GoIP supports the transparent transmission of the caller number from the PSTN to the VoIP. GoIP is designed to work in conjunction with key phone systems and IP-PBX to provide GSM communications. The extensive compatibility of the GoIP makes it an ideal choice to be deployed in multi-vendor open architecture VoIP networks. GoIP is a great way to provide fast phone service deployment where regular PSTN line may not be readily available. GSM gateway also provides significant savings in usage, infrastructure and maintenance cost compared to conventional PSTN. The GoIP features embedded SIP and H.323 protocols with flexible setting. The bi-directional password authentication (call authorization) and trust list authentication greatly minimize the risk of charge losses and the flexible routing function can meet special requirements of various call forwarding. In particular, the GoIP gateway supports multi device groups, with flexible setting of large GSM gateway groups with different channel numbers. With its low price, excellent voice quality, and powerful features, the GoIP series gateway is the first choice for system integrators, traffic operators, and softswitch manufacturers. ---- VoIP GSM Gateway GoIPbridges the GSM and the IP networks by enabling voice communications. It is ideal for VoIP to Local termination where a fixed telephone line (PSTN) is not available or for cellphone roaming via the a VoIP network. Significant savings on long distance charges can be realized. | Key Features | Open Standard VoIP Protocols (SIP&H.323) | Single or Multiple Server Registrations | Two 10/100 Ethernet for WAN / LAN connections | Peer-to-Peer IP Calls | Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer | Line Echo Cancellation | VLAN and QoS support | NAT Transversal and Router functions | Voice prompts, HTTP Web, Auto Provision support for configuration and updates | Highly stable embedded Linux operating system in high performance ARM 9 Processor | | Basic Features | LEDs for Power, Ready, Status, WAN, PC, FXS | Dial in mode or dial out mode only | Call forward from GSM to VoIP and VoIP to GSM | Dial Plan | Retransmit GSM Caller ID to VoIP terminal | | Enhanced Features | Dynamic selection of codec | Advanced jitter buffer | Automatic traversal of NAT and firewall | VLAN / Qos | Router | Echo cancellation for Speakerphone | Comfort noise generation (CNG) | Voice activity detection (VAD) | Auto provisioning (requires auto provisioning server) | On line firmware upgrade | Multi-language support: English and Chinese | | Supported Standards | ITU: H.323 V4, H.225, H.235, H.245, H.450 | RFC 1889 - RTP/RTCP | RFC 2327 -SDP | RFC 2833 - RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals | RFC 2976 - SIP INFO Method | RFC 3261 - SIP | RFC 3264 - Offer/Answer model with SDP | RFC 3515 - SIP REFER Method | RFC 3842 - A Message Summary and Message Waiting Indicator | RFC 3489 - Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs) | RFC 3891 - SIP Replaces Header | RFC 3892 - SIP Referred-By Mechanism | draft-ietf-sipping-cc-transfer-04 - Session Initiation Protocol Call Control Transfer | Codec: G.711 (A/ law), G.729A/B, G.723.1 | DTMF: RFC 2833, In-band DTMF, SIP INFO | Operating temperature: 10C to 40C (50F to 104F) | | Physical and Environmental | Storage temperature: 0C to 50C (32F to 122F) | Power: 12 VDC 500mA (110V-220V) (AC/DC adapter included) | Warranty: one year | ---- Applications: 1. Call Forward 1.Call Origination refers to a call initiated from the PSTN or cell phone netwo
