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132 * 58 graphic character dot matrix with backlight LCD, 2 SIP accounts, 3 way conference, Auto Answer, Black list, Call Center Headsets mode, Call forward, call waiting, call transfer, Call Hold, Caller ID, Dialed/Received/Missed call (each 200), Direct IP call without SIP proxy, Hot Line, Local Phonebook (upto 1000 entries), Mute,DND, Redial, Voice Mail, White list

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Anonymous call, anonymous call rejection, Call Conference, Call park, call pickup, Call Recording, DND&Call forward state synchronization, Group pickup, Intercom, paging, Music on hold

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3.2″ 462 x 278-pixel TFT LCD, Dual Gigabit Ethernet Port, PoE, WiFi, Up to 4 SIP accounts,3-way conference,Connected with up to 2x TFT LCD expansion modules (RET/AET)

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2 custom ring tone article, 8 own ring tone, 8 volume adjustable + Mute mode, Acoustic Echo Cancellation (AEC), Adaptive Jitter Buffer (AJB), Auto Gain Control (AGC), Comfort Noise Generation (CNG), Full-duplex hands-free speakerphone with AEC (Acoustic Echo Cancellation), Narrowband CODEC: G.711a/u, G.723.1, G.726-32K, G.729AB, Packet Loss Concealment (PLC), Voice Activity Detection (VAD), Wideband codec: G.722, L16,

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AES encryption for configuration file, and P2P mode, Digest authentication using MD5/MD5-sess, DTMF: In-band, HTTP / HTTPS web server, IEEE802.1X, LAN / PC: support bridge mode, LLDP, NAT transverse: STUN mode, Open VPN support, Package tracing export, Phone lock, RFC2833, SIP connection mode: Proxy mode, SIP INFO, Support DNS SRV (RFC3263), Support QoS, Syslog, TLS, User and administrator level access control, VLAN

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Atcom

Zycoo

Dinstar

Grandstream

Grandstream

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Announcements, Built in Conference Bridge/Service, Calling Queues (ACD/IVR), Dictation, Directory, Fax to Email, Feature Rich User Control Panel – Visual Voicemail, Flexible Time Based Call Routing, Follow Me/ Find Me Calling, Hunt/Ring Groups, Music On Hold, Personal IVRs, Secure Communications (SRTP/TLS), Support for Video Calling, Voice Mail Blasting, Wake Up Calls

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Call Forwarding, Call History – Call Detail Records and Call Event Logging, Call Recording, Call Screening, Call Transfer, Call Waiting, Caller Blacklisting, Caller ID, DISA, Do Not Disturb, Paging/Intercom, Speed Dials, Three Way Calling, Voice Mail – Voicemail to Email

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Open Standards Support for Multiple Signaling Protocols: SIP, IAX2, PRI/T1/E1, POTS/Analog, ISDN, Soft Phone Support, Specialty Device Support – Door Phones, Overhead Paging, Strobe Alerts, Paging Gateways, Voice Gateways, Failover Devices, WebRTC – Browser Based Calling (thru UCP)

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Bulk Import Utilities (Trunks, Extensions, Users, Phone Numbers), Integrated Intrusion Detection, System Dashboards

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User Control Panel (UCP)

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2 Port GSM Module, 2FXS-FXO Module, 4 FXO Module, 4 FXS Module, 4 Port GSM Module, Without Module

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2 Port GSM Module, 2FXS-FXO Module, 4 FXO Module, 4 FXS Module, 4 Port GSM Module, Without Module

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32 Port

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SIP V2.0 RFC3261, SDP RFC2327, Session Timer RFC4028, RTP/RTCP RFC3551, SIP Registration, SIP Trunk ( Peer Mode), SIP Trunk Group, Ringback (Immediately, Alerting), Configurable SIP Release Code, DNS SRV/A Query, Outbound Proxy, DTMF mode: Signal/RFC2833, NAT Traversal Dynamic NAT, Static NAT, STUN, G.711A/U law, G.723.1, G.729A/B, Silence Suppression & Detection, Comfort Noise Generation(CNG), Voice Activity Detection(VAD), Echo Cancellation(G.168), Adaptive (Dynamic) Jitter Buffer, Call Progress Tone Generation, Programmable Gain Control

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GXP 1450

GXP 1760

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Intuitive graphic user interface (GUI), downloadable phone book (XML,
LDAP), support for anonymous call using privacy header, MLS (multi
language support)
Voice mail indicator, downloadable custom ring-tones, call hold, call
transfer (attended/blind), call forward, call waiting, caller ID, mute, re, ,
call log, caller ID display or block, Do-Not-Disturb (DND) and volume
control3-way conference,, plan prefix,, -plan support, off-hook auto, ,
auto answer, early, and speed

Hold, transfer, forward, 5-way conference, call park, call pickup, sharedcall-appearance(SCA)/bridged-line-appearance(BLA), downloadable
phonebook(XML, LDAP, up to 500 items), call waiting, call log(up to 500
records), XML customization of screen, off-hook auto dial, auto answer, click-to-dial, flexible dial plan, hot desking, personalized music ringtones and
music on hold, server redundancy and fail-over

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Support SIP 2.0, TCP/UDP/IP, PPPoE, RTP/RTCP, SRTP by SDES, HTTP, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP, SIMPLE/PRESENCE protocols, TR-069, 802.1x
Support multiple SIP accounts and up to 11 media channels
concurrently
Support SIP PUBLISH method (RFC 3903), SIP Presence package
(RFC 3856, 3863) for use of MFKs, SIP Dialog package (RFC 4235)
Support for SIP MESSAGE method (RFC 3428)

SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP, ICMP, DNS(A record, SRV, NAPTR), DHCP, PPPoE, TELNET, TFTP, NTP, STUN, SIMPLE, LLDP, LDAP, TR-069, 802.1x, TLS, SRTP, IPV6

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User and administrator level passwords, MD5 and MD5-sess based
authentication, AES based secure configuration file, SRTP, TLS, 802.1x
media access contro

User and administrator level passwords, MD5 and MD5-sess based
authentication, 256-bit AES encrypted configuration file, SRTP, TLS, 802.1x
media access control, Kensington Security Slot (Kensington Lock) support

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2 line keys with dual-color LED and 2 independent SIP accounts, HOLD, TRANSFER, CONF, VOLUME, HEADSET, MUTE, SPEAKERPHONE, SEND/REDIAL, PHONEBOOK, MESSAGE, 3 XML
Programmable Softkeys, 5 Navigation keys

6 line keys with up to 3 SIP accounts, 4 XML programmable context sensitive
softkeys, 5 navigation/menu keys, 8 dedicated function keys for:
PHONEBOOK, TRANSFER, CONFERENCE, HEADSET, MUTE, SEND/REDIAL, SPEAKERPHONE, VOLUME

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Full-duplex hands-free speakerphone
Advanced Digital Signal Processing (DSP)
Dynamic negotiation of codec and voice payload length
Support for G.723, 1 (5.3/6.3K), G.729A/B, G.711 a/µ-law, G.726-32, G.722 (wide-band), and iLBC codecs
In-band and out-of-band DTMF (in audio, RFC2833, SIP INFO)
Silence Suppression, VAD (voice activity detection), CNG (comfort noise
generation), ANG (automatic gain control)
Acoustic Echo Cancellation (AEC) with Acoustic Gain Control (AGC) for
speakerphone mode, Support side tone
Adaptive jitter buffer control (patent-pending) and packet delay and loss
concealment
HD audio handset with HD wideband audio codecs for excellent doubletalk
performance

Yes, HD handset and speakerphone with support for wideband audio

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English, German, Italian, French, Spanish, Portuguese, Russian, Croatian, simplified and traditional Chinese, Korean, Japanese and more

English, German, Italian, French, Spanish, Portuguese, Russian, Croatian, simplified and traditional Chinese, Korean, Japanese and more

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Via keypad/LCD, Web browser, or secure (AES encrypted) central
configuration file, manual or dynamic host configuration protocol (DHCP)
network setup
Support NAT traversal using IETF STUN and Symmetric RTP
Support for IEEE 802.1p/Q tagging (VLAN), Layer 3 ToS

g Firmware upgrade via TFTP / HTTP / HTTPS, mass provisioning using TR-069
or AES encrypted XML configuration file