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Ports

1 Port, 2 Port, 4 Port, 8 Port, 16 Port

24 Port, 32 Port, 4 Port, 8 Port, 16 Port

32 Port

LEC

No, Yes

DMA

Yes

DBA

Yes

Field Firmware Upgrade

Yes

Inter Operable

Yes

Compatibility

Asterisk, Elastix, FreeBPX, FreeSWITCH, IP BRICK, trixbox, YATE

SS7 Signaling Support

YES

PRI Switch Compatibility

Euro ISDN, Network or CPE, AT&T 4ESS, CAS Voice Modes

Guarantee

25 Years from Manufacturer

3 Years From Manufacturer

N/A

Brand

Sangoma

Synway

Sangoma

Atcom

Dinstar

Grandstream

Fanvil

Yealink

PBX Interface

Connectors: 2-32 shielded female RJ-11 jacks, Number of ports: 2-32 FXO/FXS configurable, Use multiple gateway units for higher port counts

Network Interface

Connector: 2 shielded female RJ-45 jack for LAN, Network Interface: 10/100 Base-T Ethernet LAN port

VoIP Protocols

RTP/RTCP for delivery of voice, SIP per RFC 3261

FoIP Protocol

T.38 FoIP transcode fax from T.30 fax protocol (supporting V.17) modulation schemes, T.38 for transmission over a packet network

Voice Support

DTMF detection, and fax tone detection, G.711 μ-Law and A-Law, G.723.1, G.729AB Silence suppression with comfort noise G.168 automatic echo cancellation Call Progress Analysis (CPA), Including Positive Voice Detection, Positive Answering Machine Detection (PAMD)

Call Routing

From IP to PSTN or from PSTN to IP, IP fault tolerance, IP load balancing, User configuration list of VoIP endpoints

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

FXO/FXS

FXO, FXS

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

Lan

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

E1/T1 Port

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

Phone Features

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

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

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

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

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

Features

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

Telephony Features

Hold, transfer, forward (unconditional/no-answer/busy), call park/pickup, 4-way
conference, shared-call-appearance (SCA) / bridged-line-appearance (BLA), 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

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

6 VoIP accounts, One-touch speed dial, redial, Call forward, call waiting, Call transfer, call hold, Call return, group listening, Mute, auto answer, DND, 3-way conference call, Direct IP call without SIP proxy, Ring tone selection/import/delete, Hotline, emergency call, Set date time manually or automatically, Dial plan, XML Browser, Action URL/URI, Integrated Screenshots, RTCP-XR, USB port (2.0 compliant) Bluetooth earphone through BT40, Wi-Fi through WF40, USB call recording through USB flash drive, Enhanced DSS key

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

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

3 line keys with dual-color LED and 3 SIP accounts, 3 XML programmable context
sensitive soft keys, 5 (navigation, menu) keys, 8 BLF keys, 13 dedicated function
keys for MUTE, HEADSET, TRANSFER, CONFERENCE, SEND and REDIAL, SPEAKERPHONE, VOLUME, PHONEBOOK, MESSAGE, HOLD, PAGE/INTERCOM, RECORD, HOME

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:

8 line keys with LED, 8 line keys can be programmed up to 21
paperless DSS keys (3-page view), 8 features keys: message, headset, conference, mute, hold, transfer, redial, hands-free speakerphone, 4 context-sensitive “soft” keys, 6 navigation keys, 2 volume control keys, Illuminated message key, Illuminated headset key

Audio Features

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

Network and Provisioning

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

Ip Phone

GXP 1630

Fanvil X3G

SIP-T27G

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

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

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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