Please Note: The Aastra AT610 has been discontinued. For an alternative, we highly recommend the Aastra AT810P.
Atcom AT610 SIP/IAX IP Phone
The Atcom AT610 IP Phone adopts SIP protocols and multiple voice compression codec to directly convert analog voice into IP packet for internet transport, meaning it is effectively using the existing bandwidth to provide PSTN quality voice service.
Atcom AT610 SIP/IAX IP Phone Key Features
Full duplex hands-free speakerphone
Call control features
Echo cancellation
Support Voice Gain Setting
Support PPPoE for xDSL
The style of the AT610 is in accordance with other phones of Atcom AT6 series, meaning the phone is solid and reliable in a business environment. With Broadcom solution, it offers high quality voice stream and is compatible with various platforms including Asterisk, FreePBX, Broadsoft and Cisco call manager etc.
Atcom AT610 IP Phone Technical Specifications
VoIP
Support SIP 2.0 (RFC3261) 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