JXDH-6503C is a new compact (1RU only) chassis and cost-effective QAM modulator, JXDH-6503C has 1 modules, 1 1000Mbps IP inputs, 1 SPF input, the RF output supports 16QAM~256QAM modes. The JXDH-6503C has very convenient management interface, the user can complete all operation access through the Ethernet port on device.
Supplier: Digital set top boxes dvb c & dvb, digital catv headend equipments like encoders, ip streamers, edge qam modulators, edfa, etc Buyer: Electronic components
We offer state of the art, Power Semiconductors like Diodes, Thyristors, Modules, Bridge Rectifiers, Rotating Assemblies, selenium Rectifiers, Selenium Surge Suppressor, etc
Key Features of the IP Phone UTP3000
- 3 lines indicators with individual SIP account profiles
- SIP supports SIP domain, SIP authentication (none, basic, MD5), DNS name of server, Peer to Peer/ IP call
- Compatible with IAX2 protocol
- Large 128X96 high resolution graphic backlit LCD (3.2í¦)
- RJ9 and 3.5MM headset jack
- 6 programmable keys, 3 context-sensitive soft keys, a 5-position navigation key, volume keys and predefined keys for voicemail, call transfer, call hold, mute, redial, speaker, phonebook, etc.
- Full-duplex speakerphone with advanced acoustic echo cancellation (96ms max filter length).
- Dual 10/100Mbps Ethernet ports (switched/routed) with integrated Power over Ethernet (802.3af)
- Support DHCP (client/server), Static IP, PPPoE for xDSL
- Support codec: G.711(A-law/u-law), G.723.1 (5.3k/6.3k), G.729A/B, G.726, G.722
- 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.168/165), and AGC (Automatic Gain Control)
- DTMF relay: RFC2833, SIP info
- Call features: voicemail, SMS, caller ID display or block, conference call, call Forward, call Transfer (blind or attended), call hold, call waiting, paging and intercom, call park/pickup, join call, click to dial, DND, black list, limited list, call history
- Call Logs: Incoming call, Outgoing call, Missed call (100 entries each); Phonebook: 500 entries
- Support NAT Traversal (STUN); VLAN (voice VLAN / data VLAN); QoS with diffserv; VPN (L2TP); SRTP security protocol; SNTP Client; DMZ; Firewall; DNS relay; Main DNS and secondary DNS server.
- Support auto-provisioning through TFTP/TFP/HTTP for mass deployment
- Support management via keypad, web interfaces and telnet
- Reversible base stand / wall mount.
Main Features
Base on Asterisk
Configuration by Web
Built-in SIP/IAX Server
Static/DHCP/PPPoE network access
Codec: G.711-Ulaw, G.711-Alaw, G.726, G.729, GSM, SPEEX
SIP/IAX Trunk(use with VoIP Trunk operator)
Zap Trunk(Use with PSTN)
SIP/IAX Extensions(connect with IP Phone)
Zap Extensions(connect with Analog Phone)
Voice Mail Server
Flexible Dial Plan
Call Conference
IVR Server
Music On Hold
Call Queue
Call Logs
Support IP Phone with Key function
FAX T.38
Other basic function:
The S100 VoIP Phone is fully supported both H.323 and SIP standard compliant residential gateway that provides a total solution for integrating voice-data network and PSTN. By simple installation, this S100 IP Phone provides voice connectivity over the IP network and to the Public Switched Telephone Network (PSTN). Besides, it provides high voice quality and optimized packet voice streaming over managed and public (Internet) IP networks.
