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Original price was: Rs.4,720.0.Current price is: Rs.4,543.0. Inc. Tax
Original price was: Rs.10,030.0.Current price is: Rs.9,204.0. Inc. Tax
Original price was: Rs.10,030.0.Current price is: Rs.9,322.0. Inc. Tax
Original price was: Rs.5,664.0.Current price is: Rs.5,074.0. Inc. Tax
Original price was: Rs.17,700.0.Current price is: Rs.15,930.0. Inc. Tax
Phone Features

2x SIP accounts, 5-way conferencing, Action URL/URI, Call forward, call waiting, call transfer, Custom ring tones / provisioning, Dial plan per account, Direct IP call, One-touch speed dial, hotline, Redial, call return, auto answer, RTCP-XR (RFC3611), VQ-RTCPXR (RFC6035), Set date time automatically or manually

IP-PBX Features

Anonymous call, anonymous call rejection, Busy Lamp Field (BLF), Call park, call pickup, Intercom, paging, Message Waiting Indicator, Music on hold, Voicemail

Voice Codecs Features

Acoustic Echo Cancellation (AEC), Adaptive Jitter Buffer (AJB), Auto Gain Control (AGC), Codecs (iLBC, G.722, G.711(A/μ), GSM_FR, G.723, G.729AB, G.726 -32), Comfort Noise Generation (CNG), DTMF (In-band, RFC 2833, SIP INFO), Full-duplex hands-free speakerphone with AEC (Acoustic Echo Cancellation), HD voice ( HD handset, HD speaker), Packet Loss Concealment (PLC), Voice Activity Detection (VAD)

Display and Indicator

132 x 48 pixel graphical LCD with backlight, Caller ID with name, number, Illuminated LEDs for line status information, Intuitive user interface with icons and soft keys, LED for call and message waiting indication, National language selection

Feature Keys

2 line keys with LED, 4 context-sensitive “soft” keys, 6 features keys (Headset, Speaker, Hold, Mute, Transfer, Conference), 6 navigation keys, Volume control keys

Interface

1x RJ9 (4P4C) handset port, Dual-port 10/100 Mbps Ethernet, Power over Ethernet (IEEE 802.3af) (PoE), class 3

Management

Auto-provision via HTTP / HTTPS, FTP / TFTP, Auto-provision with PnP, Call hold, mute, DND, Configuration , Browser / LCD-Menu / auto-provision, HD speaker, Local tracing log export, system log, Phone lock for personal privacy protection, Reset to factory, restart, reboot

Network and Security

AES encryption for conguration le, DHCP / static / PPPoE, Digest authentication using MD5/MD5-sess, DNS-NAPTR/DNS- SRV (RFC 3263), HTTP / HTTPS web server, HTTPS certicate manager, IEEE802.1X, IPV4 / IPV6, NAT Traversal STUN mode, Open VPN, QoS( 802.1p/Q tagging (VLAN) , Layer 3 ToS DSCP), SIP server / proxy redundancy, SIP v1 (RFC2543), v2 (RFC3261), SRTP, Time and date synchronization by SNTP, TLS (Transport Layer Security)

Ip Phone

S300

GXP 1450

GXP 1780

Fanvil X3G

SIP-T29G

Brand

Sangoma

Grandstream

Grandstream

Fanvil

Yealink

Telephony Features

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 2000 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

Call out / Answer / Reject, Mute / Unmute (Microphone), Call Hold / Resume, Call Waiting, Intercom, Caller ID Display, Call Forwarding (Always/Busy/No Answer), Call Transfer (Attended/Unattended), Call Parking/Pick-up (Depending on Server), Redial, Do-Not-Disturb (Per Line / Per Phone), Auto-Answering (Per Line), Voice Message (on Server), Local 3-way Conference, Hot Line

16 VoIP accounts, Call hold, mute, DND, One-touch speed dial, hotline, Call forward, call waiting, call transfer, Group listening, SMS, emergency call, Redial, call return, auto answer, 3-way conferencing, Direct IP call without SIP proxy, Ring tone selection/import/delete, Set date time manually or automatically, Dial plan, XML Browser, Action URL/URI, Integrated screenshots, RTCP-XR, USB port (2.0 compliant) for: Bluetooth earphone through BT40, Contact synchronization through BT40, Wi-Fi through WF40, USB call recording through USB flash drive, Enhanced DSS Key

Protocols/Standards

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

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

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

Security

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

User and administrator level access control, MD5 and MD5-sess based authentication, 256-bit AES encrypted configuration file, TLS, SRTP, HTTPS, 802.1x media access control

SIP v1 (RFC2543), v2 (RFC3261), Call server redundancy supported, NAT traversal: STUN mode, Proxy mode and peer-to-peer SIP link mode, IP assignment: static/DHCP, HTTP/HTTPS web server, Time and date synchronization using SNTP, UDP/TCP/DNS-SRV(RFC 3263), QoS: 802.1p/Q tagging (VLAN), Layer 3 ToS DSCP, SRTP for voice, Transport Layer Security (TLS), HTTPS certificate manager, AES encryption for configuration file, Digest authentication using MD5/MD5-sess, OpenVPN, IEEE802.1X, IPv6, LLDP/CDP/DHCP VLAN, ICE

Feature Keys

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

8 line keys with up to 4 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

2 SIP Lines, HD Voice, POE Enabled(X3SP/X3G), Handset(HS) / Hands-free(HF) / Headset(HP) Mode (EHSsupports for Plantronics headsets), Desktop / Wall-mount Installation, Economical and Environmental Friendly Package, Industrial Standard Certifications:

10 line keys with LED, 10 line keys can be programmed up to 27 various features (3-page view), 8 features keys: message, headset, conference, hold, mute, transfer, redial, hands-free speakerphone, 4 context-sensitive “soft” keys, 6 navigation keys, Volume control keys, Illuminated headset key

Audio Features

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

Multi-language

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

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

Network and Provisioning

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

Voice Codecs

Support for G.711µ/a, G.722 (wide-band), G.723 (pending), G.726-32, G.729 A/B, in-band and out-of-band DTMF (In audio, RFC2833, SIP INFO)

Support for G.711µ/a, G.722 (wide-band), G.723 (pending), G.726-32, G.729 A/B, in-band and out-of-band DTMF (In audio, RFC2833, SIP INFO)

Compliance

FCC: Part 15 (CFR 47) Class BCE : EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3, EN60950-1RCM: AS/ACIF S004; AS/NZS CISPR22/24; AS/NZS 60950; AS/NZS 60950.1

FCC: Part 15 (CFR 47) Class BCE : EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3, EN60950-1RCM: AS/ACIF S004; AS/NZS CISPR22/24; AS/NZS 60950; AS/NZS 60950.1

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