[ใหม่] IP PHONE Atcom AT620P

ATCOM IP PHONE Model AT610P เป็น อุปกรณ์โทรศัพท์แบบ IP เหมากับการเชื่อมต่อกับ ตู้สาขา IP-PBX หรือ ใช้สำหรับบุคคล ต่อตรงกับ router เพื่อลดค่าโทรศัพท์ Ethernet support SIP or IAX2 protocols รองรับ POE  Call Frward,P2P 

AT610 VoIP phone is an entry level desktop phone terminal which adopts SIP or IAX2 protocols and multiple voice compression codec to directly convert analog voice into IP packet for internet transport, thus effectively using the existing bandwidth to provide PSTN quality voice service.

With Broadcom solution, it offers high quality voice stream , compatible with various Platforms such as Asterisk , FreePBX , Broadsoft , Cisco call manager etc.
The Style of AT610P are exactly in accordance with other phones of ATCOM AT6 series , which is close to the business environment and looks solid , reliable.

VOIP

Support SIP 2.0 (RFC3261) or IAX2 and correlative RFCs
Full duplex hands-free speakerphone
NAT transverse：support STUN client
SIP support SIP domain, SIP authentication (none, basic, MD5), DNS name of server, Peer to Peer/ IP call
SIP support 2 SIP lines. Can connect to SIP1 and SIP2 server at the same time
DTMF：Support SIP info, DTMF Relay, RFC2833
SIP application: support Call forward/ transfer/ holding/ waiting / 3 way talking/ paging and intercom/pickup/join call/click to dial/call park
Call control features: Flexible dial map, Hotline, Empty calling reject, Black list for reject authenticated call, limit call, No disturb, Caller ID
Support Phonebook 500 records
Incoming calls / Outgoing calls / Missing calls. Each support 100 records
Support conference and voice record on SIP server
10 kind of ring type
Support SRTP
Support MWI
Redundancy sip server capable.
Hotline.
Call Forward、Call transfer、Call hold、Call waiting, 3-way Talking、Pickup、Join call、Redial、Unredial、
Call Park、vport、click to dial
DND(Do Not Disturb),
Black List,Limit List
E.164 dial plan and customized dial rules
Codec：G.711 A/U Law, G.723.1, G.729a/b, G.722,G.722.1, 
Echo cancellation: Support G.168, and Hands-free can support 96ms, Hand free Speaker Phone
Support Voice Gain Setting, VAD, CNG
Tone generation and Local DTMF re-generation according with ITU-T
AGC(Auto Gain Control)
AEC(Auto Echo Cancellation)
VAD (Voice Activity Detection)
CNG(Comfort Noise Generation

Networking

WAN/LAN: Support Bridge and Router model（optional）
Support basic NAT and NAPT（optional）
Support PPPoE for xDSL
Support DHCP Client on WAN
Support DHCP Server on LAN（optional）
Support VLAN (optional: voice vlan/data vlan)
QoS with DiffServ
Support DMZ（optional）
Support VPN (L2TP/UDP TUNNEL)（optional）
Support main DNS and secondary DNS server
Support DNS Relay（optional） 
Support SNTP Client, Firewall
Network tools in telnet server: Including ping, trace route, telnet client
PoE

Protocal
MAC Address
TCP:Transmission Control Protocol 
DHCP:Dynamic Host Configuration Protocol
PPPoE:PPP Protocol over Ethernet
PoE(option)
SNTP, Simple Network Time Protocol
STUN - Simple Traversal of User Datagram ...
MD5 Message-Digest Algorithm
DNS: Domain Name Server
RTP: Real-time Transport Protocol
RTCP:Real-time Control Protocol
Telnet:Internet's remote login protocol
HTTP:Hyper Text Transfer protocol
FTP:File Transfer protocol
TFTP:Trivial File Transfer Protocol