Router SIP VoIP ATA

タイトルまたは説明

SPE gateway connects directly to phones, fax machines, key systems, PSTN lines, or a PBX. Our SPE gateway integrates voice and fax communications into your data network providing distributed IP telephony and toll bypass savings to remote offices of multi-location businesses. Model: SPE2.1(2 FXS), SPE2.1pro(2 FXS 1 FXO) FXO port allows local and remote PSTN-to-VoIP call origination and VoIP-to-PSTN call termination, automated call routing through PSTN, and power fail-over. PSTN Pass-through allows local PSTN line service for income/outbound call and failover. Product Key Features - Supports TCP/UDP/IP, RTP/RTCP, HTTP, ICMP, ARP/RARP, DNS, DHCP (both client and server), PPPoE, TFTP etc - Built-in router, NAT and Gateway - Supports SIP - Multiple(1/2/4) FXS and necessary FXO ports with independent telephone numbers - Simultaneous signaling and media encrypting and mangling - Free access to customer network - Intelligent voice routing and discovery - Interoperable with various market-leading third party Softswitch or Sip Server - Supports popular voice-coders including G.**3, G.**9AB and G.**1 (A law and U-law), FAX and Modem pass-through - Supports PSTN-Relay in case of power failure and external control - Supports standard voice features such as Caller Id, Consultation Hold, Call Waiting, Blind Call Transfer, Attended Call Transfer, Call Forward, Mid-Call DTMF, Flash, Three-way Conference - Supports Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), AGC and Line Echo Cancellation G.**8 **8ms - Supports DIGEST authentication (MD5 only) - Supports layer*3 (DiffServ, ToS) QoS - Supports automated NAT traversal without manual manipulation of firewall/NAT - Supports remote automated provisioning and software upgrade even through firewall/NAT to enable "zero configuration" and "plug-and-dial" for end users - Supports device configuration via built-in IVR by dialpad, Web browser or central configuration file