AT610 IP Phone - Broadcom Chipset Inside. 2x10/100Mbps Ethernet interfaces - compatible with various Platforms such as Asterisk , FreePBX , Broadsoft , Cisco call manager.

Features
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

Voice
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

Количество