Linksys PAP2T voip gateway
€ 49,95 Oorspronkelijke prijs was: € 49,95.€ 29,95Huidige prijs is: € 29,95. incl. 23% BTW
The Linksys Internet Phone Adapter enables high-quality feature-rich VoIP service through your broadband Internet connection . Just Plug it into your home or office Router or Gateway and use the two standard telephone ports to connect analog phones or use one of the ports for a fax machine . Each phone port operates independently , with separate phone service and phone numbers , like having two telephone lines . You’ll get clear reception and a reliable fax connection , even while using the Internet at the same time .
With Internet telephony , along with low domestic and international phone rates , an impressive array of special telephone features are avialable . Choose your preferred free local dialing area code , regardless of where you live . Or add a virtual telephone number in any area code , forwarded to your Internet phone , You can even add a tollferr number . The Linksys Internet Phone Adapter is compatible with these and of the other special telephone features that are available from your Internet telephony service provider , such as Caller ID , Call Waiting , Voicemail ,Call Forwarding , Distinctive Ring ,and much more .
Data Networking
- MAC Address (IEEE 802.3)
- IPv4-Internet Protocol v4 (RFC 791 ) upgrateale to v6 (RFC 1883)
- ARP-AddressResolution Protocol
- DNS-A Record (RFC 1706) , SRV Record (RFC 2782)
- DHCP Client-Dynamic Host Configuration Protocol (RFC 2131)
- ICMP-Internet Control Message Protocol (RFC 792)
- TCP-Transmission Control Protocol (RFC 793)
- UDP-User Datagram Protocol (RFC 768)
- RTP-Real Time Protocol (RFC 1889)
- Diffserv (RFC 2475) , Type of Service-TOS (RFC 791/1394)
- SNTP-Simple Network time protocol (RFC 2030)
Voice Gateway
- SIPv2 : Session Initiation Protocol v2 (RFC 3261 , 3262 , 3263 , 3264)
- SIP Proxy Redundancy-Dynamic via DNS SRV , A Records
- Re-registration with Primary SIP Proxy Server
- SIP Support in network Address Translation Networks-NAT (incl. STUN)
- Secure (Encrypted) Calling via Pre-Standard Implementation of Secure RTP
- Codec Name Assignment
Voice Algorithms
- G.711 (A-law and u-law)
- G.726 (16 / 24 /32 /40 kbps)
- G.729A
- G.723.1 (6.3 kbps , 5.3 kbps)
- Dynamic Payload
- Adjustment Audio Frames per Packet
Fax Capability
- Fax Tone Detection and Pass-through (Using .711)
- DTMF : In-band & Out-of-band (RFC 2833) (SIP Info)
- Flexible Dial Plan Support with Interdigit Timers and IP Dialing
- Call Progress Tone Generation
- Jitter Buffer-Adaptive
- Frame Loss Concealment
- Full Duplex Audio
- Echo Cancellation (G.165 /G.168)
- VAD-Voice Activity Detection with Silence Suppression
- Attenuation / Gain Silence Suppression
- Flash Hook Timer
- MWI-Message Waiting Indicator Tone
- VMWI-Visual Message Waiting Indicator via FSK
- Polarity Control
- Hook Flash Event Signaling
- Caller ID Generation (Name & Number)-Bellcore , DTMF ,ETSI
- Music on Hold Client
- Streaming Audio Server-up to 10 sessions
Provisioning , Administration & Maintenance :
- Web Browser Administration & Configuration via Integrated Web Server
- Telephone Key Pad Configuration with Interactive Voice Prompts
- Automted Provisioning & Update Availability via SIP Notify
- Non-intrusive , In-Service Upgrades
- Report Generation & Event Logging
- Stats in BYE Message
- Syslog & Debug Server Records-Per Line Configurable
Refurbished
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