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Rs.82,600.0 From: Rs.70,800.0 Inc. Tax
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Original price was: Rs.29,500.0.Current price is: Rs.24,190.0. Inc. Tax
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Original price was: Rs.20,650.0.Current price is: Rs.17,936.0. Inc. Tax
E1/T1 Port

1E1/T1 and 30 SIP channels, 2E1/T1 and 60 SIP channels, 4E1/T1 and 120 SIP channels

1E1/T1 and 30 SIP channels, 2E1/T1 and 60 SIP channels, 4E1/T1 and 120 SIP channels

IP Bearer Features

Coder support: G.711A,G.711U, G.729 A/B,GSM, iLBC, RFC 2833,RF 3261,SIPINFO,INBOUND, Comfort noise generation, Compliant with TCP/UDP, HTTP, ARP/RARP,DNS,NTP,TFTP,TELNET,STUN and more IP protocols, Digit transmission via RFC 2833 (SIP), Echo cancellation: G.168 128 ms tail length, Hosted NAT, T.38 real-time fax, T.38 – G.711 interworking, Voice activity detection and packet loss concealment

OAM&P

Network Time Protocol (NTP), SNMP MIBs, Web User Interface (WebUI) supports configuration via browser

Power Requirements

AC Power Supply Range 100 – 240 VAC, The power supply will operate at frequencies between 47 Hz and 63 Hz

Capacity

30 – 120 TDM channels per 1U shelf, 30 – 120 VoIP channels per 1U shelf

IP Interfaces

IP Dual redundant 2 – 100/1000 Base-T Ethernet for VoIP payload and signaling

TDM Signaling Protocols

4 SS7 links in standalone configuration, ISDN PRI, ISDN/SS7, SS7 ISUP, SS7 TCAP

IP Protocols

Core SIP Specifications and Notable Extensions, RFC 3261 SIP Basic, RFC 3262 SIP PRACK, RFC 3265 SIP Subscribe/Notify

Notable SIP Extensions

IP and ISUP interworking and more, RFC 3398 ISUP/SIP Mapping, RFC 3711 SRTP (for SIP), Tel URI – RFC 3966

Guarantee

3 Years From Manufacturer

N/A

Brand

Synway

Sangoma

Sangoma

Fanvil

Atcom

Grandstream

Grandstream

Fanvil

Fanvil

Call Routing

Dial planner – sophisticated call routing capabilities, standalone or gatekeeper/proxy integration, Direct Dialing In (DDI), NAT traversal, SIP registration to multiple proxies

Dial planner – sophisticated call routing capabilities, standalone or gatekeeper/proxy integration, Direct Dialing In (DDI), SIP registration to multiple proxies

Audio Encoding & Decoding

G.711A 64 kbps, G.711U 64 kbps, G.723 5.3/6.3 kbps, G.726, G.729a (8kbps), T.38

G.711A 64 kbps, G.711U 64 kbps, G.723 5.3/6.3 kbps, G.726, G.729a (8kbps), T.38

Lan

Amount: 2 (10/100 BASE-TX (RJ-45)), full / half duplex

1000BaseT / 100BaseTx / 10BaseT, full / half duplex, 1x RJ-45 Gigabit Ethernet

Ports

24 Port, 50 Port, 8 Port

FXO/FXS

FXO, FXS

Telephony Interface

24 FXS ports on an RJ-21 connector, 600R, 900R or CTR-21 line impedance

Calling Features

Message waiting indicator – audible, visual, Call forward – unconditional, busy, no-answer, Call transfer – blind, consultative, Call Waiting, Caller ID, Do Not Disturb, Music on hold, Three Way Calling

Phone Features

Android 4.2 OS, Supports 6 SIP servers and Backup SIP proxy servers., Supports Acoustic echo cancellation (AEC) – 128ms max filter length in duplex speaker phone mode, Supports HDMI, Supports high quality video call, Supports PLC & adaptive jitter buffer, Supports SIP 2.0 (RFC3261) and correlative RFCs, Supports SIP UDP/TCP/TLS, Supports USB Host

1 SIP account, 132 * 52 graphic character dot matrix with backlight LCD, Black/White List Call Filtering, Call Center Headsets mode, Call Forward, Call Transfer, Call Waiting, Hotline, Call Hold, Auto Answer, Caller ID, Redial, Mute, DND, Dialed/Received/Missed call (each 200), Direct IP call, Local 3-Way Conferencing, Local Phonebook (upto 1000 entries), Speed Dial, Voice Mail

IP-PBX Features

Auto Answer (Hands- free / Headset), Auto Redial / un-redial, Barring function for outgoing calls, Black List, BLF List, Call Completion, Call Conference, Call Forwarding, Call Holding, Call Rejection, Call Transfer (blind/attended/alert), Call Waiting, Caller ID Display, Capable of 10 way conversation, Dial without registration, Direct IP call without SIP proxy, Do Not Disturb, Flexible dial plan, Headset ring, Hot desk function, Hotline/Warm-line, Intercom/Intercom barge, Join Call, Password dial, Pickup, Soft DSS Key (Upto 100), Support messaging and MWI, Support multi line and pre-dial, Supports Call Logs with Missed / Incoming / Outgoing calls, Voice recording during talking /local

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

Display and Indicator

Bandwidth selection: 64kbps~4Mbps, Camera- Adjustable Position, Frame rate selection: 10~30fps, Image codec: JPEG/PNG/BMP/GIF, LCD Size Diagonal:7 inch (800 x 480) Capacitive touch screen, SD Interface – TF Card Support Upto 32G, Supports up to 4 video display mode, Video call resolution: QCIF / CIF / VGA / 4CIF (1280x720P Optional), Video codec: H.264 / H.263, Video format: MP4/3GP/FLV

132 x 52 monochrome LCD with backlitDual Ethernet Port, PoESingle SIP account,3-way conference, Wall-mount and Desktop bracket design (No Power adapter included)

Voice Codecs Features

Adaptive Jitter Buffer (AJB), Audio format: WAV/MP3/OGG, Comfort Noise Generation(CNG), DTMF: In-band, Out-of-Band – DTMF-Relay (RFC2833) / SIP INFO, Full-duplex hands-free speaker phone with AEC (Max filter length – 128ms), Narrowband codec: G.711(A/μ), G.723.1, G.729AB, iLBC, AMR, Packet Loss Concealment (PLC), Voice Activity Detection (VAD), Wideband CODEC: G.722

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,

Network and Security

DHCP/ static/ PPPoE, STUN, Supports main DNS and secondary DNS server, Supports MD5 authentication, Supports PoE (802.3af), Supports QoS: 802.1p/q, DSCP, Supports SIP SRTP,, Supports SNTP Client, Supports VLAN, Supports VPN L2TP / PPTP / IPSec, Supports Web Filter, Supports Web HTTP / HTTPS, WAN/LAN: support bridge mode

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

Ip Phone

GXP 1625

GXP 2120

Fanvil X1

Fanvil X6

Telephony Features

Hold, transfer, forward (unconditional/no-answer/busy), 3-way conferencing, call park/pickup, downloadable phone book (XML, LDAP, up to 500
items), call waiting, call history (up to 200 records), off-hook auto dial, auto
answer, click-to-dial, flexible dial plan, hot desking, personalized music
ringtones, server redundancy & fail-over

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, redial, call log, caller ID
display or block, Do-Not-Disturb (DND) and volume control
Multi-party conferencing (up to 5), dial plan prefix, off-hook auto dial, auto answer, early dial and speed dial (on some models)

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

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

Protocols/Standards

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

Support SIP 2.0, TCP/UDP/IP, PPPoE, RTP/RTCP, SRTP by SDES, HTTP, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP, SIMPLE/PRESENCE protocols
Supports multiple SIP accounts
Supports SIP PUBLISH method (RFC 3903), SIP Presence package (RFC 3856, 3863) for use of 7 MFKs, SIP Dialog package (RFC 4235)
Supports SIP MESSAGE method (RFC 3428)
Stores up to 100 incoming IM messages

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

DIGEST authentication and encryption using MD5 and MD5-sess, SRTP, SIP over
TLS (pending)

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

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

Feature Keys

2 line keys with dual-color LED and 2 SIP accounts. 3 XML programmable
context sensitive soft keys. 5 (navigation, menu) keys. 13 dedicated function
keys for MUTE, HEADSET, TRANSFER, CONFERENCE, SEND and
REDIAL, SPEAKERPHONE, VOLUME

6 line keys with dual color LED and 6 independent SIP accounts, HOLD, TRANSFER, CONF, VOLUME, HEADSET, MUTE, SPEAKERPHONE, SEND/REDIAL, PHONEBOOK, MESSAGE, 3 XML
Programmable Softkeys, 5 Navigation keys

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:

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

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)

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)

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

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

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

Audio Features

Full-duplex hands-free speakerphone, headset enabled
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 (wideband), GSM 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), AGC (automatic gain control)
Acoustic Echo Cancellation (AEC) with Automatic Gain Control (AGC) for
speakerphone mode, Support side tone
Adaptive jitter buffer control and packet delay & loss concealment

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

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