Gateway VoIP fxo 24 porte

Gateway fxo 24 porte Openvox MAG1000-24O

VoIP gateway fxo 24 porte Openvox MAG1000

Il gateway fxo è utilizzabile nei casi in cui si debbano convertire linee o interni analogici in VoIP.

Un utilizzo tipico è quello della conversione di interni analogici di un centralino telefonico in VoIP per essere poi riconvertiti  a valle in analogico (sfruttando gateway fxs).

Altra applicazione: installare linee esterne analogiche su un centralino IP (che supporta quindi solo linee SIP - Trunk SIP).

Performance and Features
● 16/ 24/ 32 FXO/FXS ports
● Stable function, flexible dialplan and friendly Web GUI
● Strong security and high stability

Overview
OpenVox MAG 1000 is a multifunctional analog gateway for seamless connection between IPPBX, fax machines, analog phones and operators. It provides three interfaces of RJ11/RJ45/RJ21, covering almost all the wiring needs of users in all scenarios. In addition, it has excellent full concurrent voice/ fax pro-cessing capabilities, strong performance and high stability, and provides high-quality call services for oper-ators, enterprises, call centers and residential users in residential communities.
The MAG1000 Analog Gateways are developed for interconnecting a wide selection of codecs including G.711A, G.711U, G.729A, G.722, G.726, iLBC. MAG1000 series use standard SIP protocol and compati-ble with Leading VoIP platform, IPPBX and SIP servers, such as Asterisk, Issabel, 3CX, FreeS-WITCH ,BroadSoft and VOS VoIP operating platform.
Features

System Features
● NTP time synchronization and client time synchronization
● Support modify username and password for web login
● Update firmware online, backup/restore configuration file
● Abundant Log Info, Automatically Reboot, Call status display
● Language selection (Chinese/English)
● Open API interface, support the secondary development of applications
● Support SSH remote operation and restore the factory settings
● Support center management system
● Auto Provisioning, Backup/Restore
● Syslog: Debug, error, alert and CDR

Telephony Features
● Support Volume adjustment, Gain adjustment, call transfer, call hold, call wait-ing, call forward, Caller ID display
● Three way calling, Call transfer, Dial-up matching table
● Support T.38 fax relay and T.30 fax transparent, FSK and DTMF signaling
● Support T.38/Pass-through, up to 14.4Kbps
● Support Ring cadence and frequency setting, WMI (Message Waiting Indicator)
● Support Echo cancellation(G.168) , Adaptive (Dynamic) Jitter buffer
● Support customizable DISA and other applications
● Comfort Noise Generation (CNG), Voice Activity Detection (VAD)
● Hook Flash
● DTMF/FSK (caller ID)
● IMS/3GPP/UDP/TCP
● SDP (RFC 2327) - Session
● ETC (3GPP TS 24.629, RFC 3515, RFC 3891, RFC 3892)
● RTP/RTCP (RFC 1889, 1890), STUN (RFC 3489)
● Outbound Proxy, QoS, Traffic shaping
● Offer/Answer (RFC3265)
● Voice/ Management VlAN, VLAN 802.1P/802.1Q
● Sync timer

SIP Features
● Support add, modify & delete SIP Accounts ,batch add, modify & delete SIP Ac-counts
● Support multiple SIP registrations : Anonymous, Endpoint registers with this gate-way, This gateway registers with the endpoint
● SIP accounts can be registered to multiple servers
● SIP V2.0 (RFC 3261, 3262, 3264)
● SIP Session Timer (RFC 4028)
● Support Primary and Backup SIP Server
● Codec G.711A/U law, G.729A/B, G726-32, G.722, iLBC
● DTMF mode: Signal/ RFC2833/ Inband
● Designed for almost all kinds of users
● Fully Compatible with Leading SIP based VoIP Telephony System
● Support voice, fax and POS terminal access


Network
● Network type：Static IP, Dynamic
● Support DDNS, DNS, DHCP, DTMF relay, NAT
● Telnet, HTTP, HTTPS, SSH , TFTP, FTP
● VPN clients：PPTP/OpenVPN/Zerotier/L2TP/N2N（Not available for Russia）
● Description Protocol (RFC 2327)
● Support IPv4 and IPv6
● SNMP/TR069

System Features
● NTP time synchronization and client time synchronization
● Support modify username and password for web login
● Update firmware online, backup/restore configuration file
● Abundant Log Info, Automatically Reboot, Call status display
● Language selection (Chinese/English)
● Open API interface, support the secondary development of applications
● Support SSH remote operation and restore the factory settings
● Support center management system
● Auto Provisioning, Backup/Restore
● Syslog: Debug, error, alert and CDR

Telephony Features
● Support Volume adjustment, Gain adjustment, call transfer, call hold, call wait-ing, call forward, Caller ID display
● Three way calling, Call transfer, Dial-up matching table
● Support T.38 fax relay and T.30 fax transparent, FSK and DTMF signaling
● Support T.38/Pass-through, up to 14.4Kbps
● Support Ring cadence and frequency setting, WMI (Message Waiting Indicator)
● Support Echo cancellation(G.168) , Adaptive (Dynamic) Jitter buffer
● Support customizable DISA and other applications
● Comfort Noise Generation (CNG), Voice Activity Detection (VAD)
● Hook Flash
● DTMF/FSK (caller ID)
● IMS/3GPP/UDP/TCP
● SDP (RFC 2327) - Session
● ETC (3GPP TS 24.629, RFC 3515, RFC 3891, RFC 3892)
● RTP/RTCP (RFC 1889, 1890), STUN (RFC 3489)
● Outbound Proxy, QoS, Traffic shaping
● Offer/Answer (RFC3265)
● Voice/ Management VlAN, VLAN 802.1P/802.1Q
● Sync timer

SIP Features
● Support add, modify & delete SIP Accounts ,batch add, modify & delete SIP Ac-counts
● Support multiple SIP registrations : Anonymous, Endpoint registers with this gate-way, This gateway registers with the endpoint
● SIP accounts can be registered to multiple servers
● SIP V2.0 (RFC 3261, 3262, 3264)
● SIP Session Timer (RFC 4028)
● Support Primary and Backup SIP Server
● Codec G.711A/U law, G.729A/B, G726-32, G.722, iLBC
● DTMF mode: Signal/ RFC2833/ Inband

link al manuale in inglese

https://help.openvox.cn/docs/analog-gateway/user-manual/mag-series-analog-gateway/