+!! ~~blue: 32 port gsm gateway ~~ |
+!![ http://www.hybertone.com/en/pro_detail.asp?proid=56| 32 port gsm gateway for call termination, HyberTone Technology] |
+{attachment id=4584} |
+ |
+!! ~~blue: Product name: 32 port gsm gateway ~~ |
+!! ~~blue: Model: GOIP32 ~~ |
+ |
+{| |
+| {attachment id=4581} |
+|} |
+ |
+!Specification |
+ |
+!! __Basic Features__: |
+ ** For call termination (VoIP to GSM) and origination (GSM to VoIP) } |
+ ** Standard SIP & H.323 protocol |
+ ** GSM: quad-band 850/900/1800/1900MHz |
+ |
+!!__Major Advantages__: |
+ ** Lightweight and Portable |
+ ** IMEI Changeable |
+** GSM Base Station Optional |
+ ** Support SIM Bank/ SIM Sever |
+ ** Manual/ Automatic Selection Operators |
+ ** Sending and Receiving SMS and USSD (Web Interface) |
+ |
+!!__Free Software/ Sever__: |
+**__Remote Control Server__: Access Interface Remotely |
+ |
+**__Relay Server__: Relay Encryption (Make Terminals Traversal the NAT without STUN and Outbound Proxy ) |
+ |
+**__SMS Server__: |
+ *** Send Bulk SMS |
+ *** Provide CDR and ASR |
+ *** Auto Balance and Recharge |
+ |
+**__SIM Server__: |
+ *** Rotate SIM Cards on Duty |
+ *** Set GSM Group (Assign several SIMs Per GSM Port) |
+ ***Set Talk Time per SIM, Set Day of week, Set Time Range |
+ *** Monitor CDR, ASR, ACD |
+ |
+!!__Key Features__ |
+ |
+ **Provide 32 cellular channels for IP-PBX |
+ **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 |
+ **Quad band GSM module: 850MHz, 900 MHz, 1800 MHz, 1900MHz |
+ **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 |
+ |
+ |
+!!__Enhanced Features__ |
+ |
+ **LEDs for Power, Ready, Status, WAN, PC, GSM |
+ **Dial in mode or dial out mode only |
+ **Call forward from GSM to VoIP and VoIP to GSM |
+ **Dial Plan |
+ **Password protection for both GSM dial in or dial out |
+ **Retransmit GSM Caller ID to VoIP terminal |
+ **Dynamic selection of codec |
+ **Advanced jitter buffer |
+ **Automatic traversal of NAT and firewall |
+ **VLAN / Qos |
+ **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 (STUN)- 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 |
+ **Web-base Management |
+ **PPP over Ethernet (PPPoE) |
+ **PPP Authentication Protocol (PAP) |
+ **Internet Control Message Protocol (ICMP) |
+ **TFTP Client |
+ **Hyper Text Transfer Protocol (HTTP) |
+ **Dynamic Host Configuration Protocol (DHCP) |
+ **User account authentication using MD5 |
+ |
+ |
+!!Lead Sales: Leo |
+!!Website: [www.hybertone.com] |
+!!Email: Leo@hybertone.com |
+!!Skype: leohybertone |
+!!Tel: +86 15118089575 |
+ |
+!!-= Facebook =- [http://www.facebook.com/HyberTone.VoIP] |
+!!-= Youtube =- [http://www.youtube.com/HyberTone] |
|