Analog Phone Adapter (ATA) UTA 2011 SIP / IAX - With Conference Call Feature Dual 10 / 100 Mbps Ethernet Ports (Switched / Routed)

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Verkäufer: voice7uk ✉️ (7.560) 98.8%, Artikelstandort: Manchester, GB, Versand nach: GB, Artikelnummer: 131730386744 Analog Phone Adapter (ATA) UTA 2011 SIP / IAX - With Conference Call Feature.

VoIP Adapter UTA2011 SIP/IAX2 VoIP Adapter with 1 FXS, 1 PSTN Pass-Through, 2 RJ45: The UTA2011 is an Analog Telephone Adapter based on SIP & IAX2 standards, offers 1 FXS port to connect existing analog telephone or fax machine to IP-based data networks and 1 PSTN Pass-Through port.
Cost-effective, easy-to-install and simple-to-use, the VoIP adapter UTA2011 converts standard telephones to IP-based networks with these benefits. Equipped with 1 FXS port, the UTA2011 can save installation cost and extend your past investment in telephones, video conferencing and speaker. With the features of dual 10/100Mbps auto-sensing Ethernet ports, DHCP (client/server), T.38 fax transmission over IP network, auto-provisioning through TFTP/FTP/HTTP server, compatible with VPN (L2TP), VLAN (voice VLAN/data VLAN), QoS with diffserv, customized dial plan, caller ID display, call hold, call waiting, call transfer, call forward, DND, IP telephony service providers and enterprise users can offer residential and business users traditional and enhanced communication services via the customers broadband connection to the Internet or Local Area Network (LAN) by the VoIP adapter UTA2011. 
The UTA2011 is an ideal VoIP telephone adapter solution that will help VoIP service providers or resellers with all of the required attributes to grow and retain customers by delivering a robust and consistent VoIP solution 

Key Features of the VoIP Adapter UTA2011
  • Support 2 SIP accounts registering simultaneously; Compatible with IAX2 protocol 
  • SIP supports SIP domain, SIP authentication (none, basic, MD5), DNS name of server, Peer to Peer/ IP call 
  • Dual 10/100Mbps Ethernet ports (switched/routed) 
  • DHCP (client/server), Static IP, PPPoE (for ADSL, Cable modem) 
  • Single FXS Port (for phone/fax machine connection) 
  • 1 PSTN Pass-Through Port (for PSTN connection) 
  • Support codec: G.711 (A-law/u-law), G.729A/B, G.726, iLBC 
  • Advanced Digital Signal Processing (DSP) to ensure superb hi-fidelity audio quality 
  • Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control); Adaptive Jitter Buffer 
  • DTMF relay: RFC2833, SIP info 
  • Support customized dial peer 
  • Caller ID with name/number 
  • Call features: Call Hold, Call Waiting, Call Forwarding (No answer/Busy/All), Call Transfer (blind/attended), Conference Call, Black List & Limited List, Do Not Disturb 
  • Hotline calling 
  • Support remote auto-provisioning through TFTP/FTP server for mass deployment 
  • Support device configuration via built-in IVR, Web browser or central configuration file 
  • Support NAT Traversal (STUN); VLAN (voice VLAN/ data VLAN); QoS with diffserv; VPN (L2TP); DMZ; Firewall; DNS relay
  Specifications of the VoIP Adapter UTA2011
Product
Description SIP/IAX VoIP Adapter with 1 FXS, 1 PSTN Pass-Through, 2 RJ45
Model UTA2011
Hardware
WAN Port
(for connecting to Internet)
1 X 10/100Mpbs RJ45 port
LAN Port
(for connecting to PC)
1 X 10/100Mpbs RJ45 port
FXS (for phone/fax connection) 1 X RJ11 port
PSTN Pass-Through 1 X RJ11 port
Memory SDRAM: 8Mpbs
Flash Memory: 2Mpbs
Features & Benefits
Standard SIP v1   (FRC2543), v2 (RFC3261)
Support 2 SIP lines
Support IAX2 protocol
SIP supports SIP domain, SIP authentication (none, basic, MD5), DNS name of server, Peer to Peer/ IP call
Voice Codec G.729A/B, G.711 (A-law/ µ -law), G.726, iLBC
Voice Standard DTMF relay: RFC2833, SIP info
Auto Gain Control (AGC)
G.168/165 compliant 16ms echo cancellation
Auto Echo Cancellation (AEC)
Voice Activity Detection (VAD)
Comfort Noise Generation (CNG)
Adaptive Jitter Buffer
Call Features Call hold, call waiting, call transfer (blind/attended), call forward, 3-way conferencing
Customized dial peer
Caller ID with name or number
DND (do not disturb), Black List, Limited List
Hotline
Peer to Peer/ IP call
Network & Management
Access Mode DHCP (client/server), Static IP, PPPoE (for ADSL, Cable modem)
Management Built-in IVR, Web browser, Telnet
Auto-provisioning through TFTP/FTP/HTTP
Firmware upgrade through FTP, TFTP
Configuration file download/upload
Support Syslog
Protocols TCP/IP/UDP, DHCP, PPPoE, SNTP, DNS, RTP, RTCP, Telnet, HTTP, FTP, TFTP
Applications NAT Traversal (STUN); VLAN (voice VLAN/data VLAN); QoS with diffserv; VPN (L2TP); DMZ; Firewall
Operating Requirements
Operating Temp. 0~50 degree C
Storage Temp. -30~65 degree C
Operating Humidity 10~95% Non-condensing
Storage Humidity 10~95% Non-condensing
Power Requirement Input 100~240V AC, Output 12V DC 450mA
Power Consumption Idle: 1.8W Active: 2.4W
Regulatory Compliance CE, FCC part 15 class B, RoHS
Packages Contents
UTA2011 ATA unit 1
Power Adapter   1  
Manual CD 1

  • Condition: Neu
  • Brand: Voice 7
  • Model: UTA -2011
  • Type: Analog Phone Adapter
  • Features: Call Hold, Call Waiting, Call Forwarding (No answer/Busy/All)
  • Interface: Ethernet (RJ-45)
  • Colour: Black
  • MPN: UTA -2011
  • Unit Quantity: 1
  • Custom Bundle: No
  • Device Type: VoIP / Internet Telephones
  • Other Features: Call Transfer (blind/attended), Conference Call, Black List etc

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