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Computer Network Accessories
Friday, 2 June
Search for product:
 IP PBX
#05989

VoIP IP PBX, 2xFXO, 2xFXS (Grandstream UCM6202)

Designed to provide a centralized solution for the communication needs of businesses, the UCM6200 series IP PBX
appliance combines enterprise-grade voice, video, data, and mobility features in an easy-to-manage solution. This
IP PBX series allows businesses to unify multiple communication technologies, such as voice, video calling, video
conferencing, video surveillance, data tools, mobility options and facility access management onto one common
network that can be managed and/or accessed remotely. The secure and reliable UCM6200 series delivers enterprisegrade features without any licensing fees, costs-per-feature or recurring fees.
Analog Telephone FXS Ports 2 ports (both with lifeline capability in case of power outage)
PSTN Line FXO Ports 2 ports (UCM6202)
Network Interfaces Dual Gigabit RJ45 ports with integrated PoE Plus (IEEE 802.3at-2009)
NAT Router Yes (supports router mode and switch mode)
Peripheral Ports USB,
SD
LED Indicators Power/Ready,
Network,
PSTN Line,
USB,
SD
LCD Display 128x32 graphic LCD with DOWN & OK button
Voice-over-Packet Capabilities LEC with NLP Packetized Voice Protocol Unit,
128ms-tail-length carrier grade Line Echo Cancellation, Dynamic Jitter Buffer,
Modem detection & auto-switch to G.711
Voice and Fax Codecs G.711 A-law/U-law,
G.722,
G.723.1 5.3K/6.3K,
G.726,
G.729A/B,
iLBC,
GSM,
AAL2-G.726-32,
ADPCM; T.38
Video Codecs H.264,
H.263,
H263+
QoS Layer 3 QoS,
Layer 2 QoS
DTMF Methods In Audio,
RFC2833,
and SIP INFO
Provisioning Protocol & Plug-and-Play TFTP/HTTP/HTTPS,
auto-discovery & auto-provisioning of Grandstream IP endpoints via ZeroConfig (DHCP Option 66 multicast SIP SUBSCRIBE mDNS),
eventlist between local and remote trunk
Network Protocols TCP/UDP/IP,
RTP/RTCP,
ICMP,
ARP,
DNS,
DDNS,
DHCP,
NTP,
TFTP,
SSH,
HTTP/HTTPS,
PPPoE,
SIP (RFC3261),
STUN,
SRTP,
TLS,
LDAP
Disconnect Methods Call Progress Tone,
Polarity Reversal,
Hook Flash Timing,
Loop Current Disconnect,
Busy Tone
Media Encryption SRTP,
TLS,
HTTPS,
SSH
Caller ID Bellcore/Telcordia,
ETSI-FSK,
ETSI-DTMF,
SIN 227 – BT
Polarity Reversal/Wink Yes,
with enable/disable option upon call establishment and termination
Call Center Multiple configurable call queues,
automatic call distribution (ACD) based on agent skills/availability/busy level,
in-queue announcement
Customizable Auto Attendant Up to 5 layers of IVR (Interactive Voice Response)
Maximum Call Capacity -Registered SIP devices- supports up to 500 registered SIP devices/users,
-Concurrent SIP calls- Up to 50 (UCM6202),
or 66% performance if calls are SRTP encrypted
Conference Bridges Up to 3,
password-protected conference bridges allowing up to 25 simultaneous PSTN or IP participants
Call Features Call park,
call forward,
call transfer,
DND,
ring/hunt group,
paging/intercom etc.
Mounting Wall mount & Desktop
Power Output 12VDC,
1.5A; Input 100 ~ 240VAC,
50 ~ 60Hz
Weight Unit weight 0.51kg,
Package weight 0.94kg
Environmental Operating 32 ~ 104ºF / 0 ~ 40ºC,
10 ~ 90% (non-condensing); Storage 14 ~ 140ºF / -10 ~ 60ºC
Dimensions 226mm L x 155mm W x 34.5mm H
Certificates FCC Part 15 (CFR 47) Class B,
Part 68,
CE EN55022 Class B,
EN55024,
EN61000-3-2,
EN61000-3-3,
EN60950-1,
TBR21,
RoHS,
A-TICK AS/NZS CISPR 22 Class B,
AS/NZS CISPR 24,
AS/NZS 60950,
AS/ACIF S002,
ITU-T K.21 (Basic Level),
UL 60950 (power adapter)
Manufacturer Grandstream
GRANDSTREAM
VoIP IP PBX, 2xFXO, 2xFXS (Grandstream UCM6202)
* For special orders only
Net Price: 352.00 USD  Unit: pcs  VAT: 23%

#05994

VoIP IP PBX, 2xFXO, 2xFXS, T1/E1/J1 (Grandstream UCM6510)

The UCM6510 IP PBX appliance is designed to bring leading edge voice, video, data, and mobility features to
enterprises, small and medium businesses, retail and residential environments in an easy-to-manage fashion. This
enterprise-grade on premise IP PBX supports E1, T1 and J1 networks and offers scalability by supporting up to 2000
users. The UCM6510 sports a 1GHz quad-core Cortex A9 processor, 1GB RAM and 32GB flash. This secure and reliable
IP PBX delivers unified communication features at an unprecedented price point without any licensing fees, costs-perfeature, or recurring fees.
Analog Telephone FXS Ports 2x RJ11 ports (both with lifeline capability in case of power outage)
PSTN Line FXO Ports 2x RJ11 ports (both with lifeline capability in case of power outage)
T1/E1/J1 Interface 1x RJ45 port
Network Interfaces Dual Gigabit ports (switched or routed) with PoE+
NAT Router Yes (supports router mode and switch mode)
Peripheral Ports USB,
SD
LED Indicators Power 1/2,
PoE,
USB,
SD,
T1/E1/J1,
FXS 1/2,
FXO 1/2,
LAN,
WAN
LCD Display 128x32 graphic LCD with DOWN & OK button
Voice-over-Packet Capabilities LEC with NLP Packetized Voice Protocol Unit,
128ms-tail-length carrier grade Line Echo Cancellation, Dynamic Jitter Buffer,
Modem detection & auto-switch to G.711
Voice and Fax Codecs G.711 A-law/U-law,
G.722,
G.723.1 5.3K/6.3K,
G.726,
G.729A/B,
iLBC,
GSM,
AAL2-G.726-32,
ADPCM; T.38
Video Codecs H.264,
H.263,
H263+
QoS Layer 3 QoS,
Layer 2 QoS
DTMF Methods In Audio,
RFC2833,
and SIP INFO
Digital Signaling TPRI,
SS7,
MFC/R2,
RBS (pending)
Provisioning Protocol & Plug-and-Play TFTP/HTTP/HTTPS,
auto-discovery & auto-provisioning of Grandstream IP endpoints via ZeroConfig (DHCP Option 66 multicast SIP SUBSCRIBE mDNS),
eventlist between local and remote trunk
Network Protocols TCP/UDP/IP,
RTP/RTCP,
ICMP,
ARP,
DNS,
DDNS,
DHCP,
NTP,
TFTP,
SSH,
HTTP/HTTPS,
PPPoE,
SIP (RFC3261),
STUN,
SRTP,
TLS,
LDAP,
HDLC,
HDLC-ETH,
PPP,
Frame Relay (pending)
Disconnect Methods Call Progress Tone,
Polarity Reversal,
Hook Flash Timing,
Loop Current Disconnect,
Busy Tone
Media Encryption SRTP,
TLS,
HTTPS,
SSH
Advanced Defense Fail2ban,
alert events,
Whitelist,
Blacklist,
strong password based access control
Caller ID Bellcore/Telcordia,
ETSI-FSK,
ETSI-DTMF,
SIN 227 – BT
Polarity Reversal/Wink Yes,
with enable/disable option upon call establishment and termination
Call Center Multiple configurable call queues,
automatic call distribution (ACD) based on agent skills/availability/work-load,
in-queue announcement
Customizable Auto Attendant Up to 5 layers of IVR (Interactive Voice Response)
Maximum Call Capacity Up to 2000 registered SIP endpoints,
up to 200 concurrent calls
Conference Bridges Up to 8 bridges,
up to 64 simultaneous conference attendees
Call Features Call park,
call forward,
call transfer,
DND,
DISA,
ring group,
pickup group,
blacklist,
paging/intercom etc.
Mounting Rack mount & Desktop
Power Input 100 ~ 240VAC,
50/60Hz; Output DC+12V,
1.5A
Weight Unit Weight 2.165 kg,
Package Weight 3.012 kg
Environmental Operating 32 ~ 113ºF / 0 ~ 45ºC,
10 ~ 90% (non-condensing); Storage 14 ~ 140ºF / -10 ~ 60ºC
Dimensions 440mm L x 185mm W x 44mm H
Certificates FCC- Part 15 (CFR 47) Class B,
Part 68,
CE- EN55022 Class B,
EN55024,
EN61000-3-2,
EN61000-3-3,
EN60950-1,
TBR21,
RoHS,
RCM- AS/NZS CISPR 22,
AS/NZS CISPR 24,
AS/NZS 60950,
AS/ACIF S002,
ITU-T K.21 (Basic Level),
UL 60950 (power adapter),
T1- TIA-968-B Section 5.2.4,
E1- TBR4/TBR12/TBR13,
E1- AS/ACIF
Manufacturer Grandstream
GRANDSTREAM
VoIP IP PBX, 2xFXO, 2xFXS,  T1/E1/J1 (Grandstream UCM6510)
* For special orders only
Net Price: 1 240.00 USD  Unit: pcs  VAT: 23%

#05990

VoIP IP PBX, 4xFXO, 2xFXS (Grandstream UCM6204)

Designed to provide a centralized solution for the communication needs of businesses, the UCM6200 series IP PBX
appliance combines enterprise-grade voice, video, data, and mobility features in an easy-to-manage solution. This
IP PBX series allows businesses to unify multiple communication technologies, such as voice, video calling, video
conferencing, video surveillance, data tools, mobility options and facility access management onto one common
network that can be managed and/or accessed remotely. The secure and reliable UCM6200 series delivers enterprisegrade features without any licensing fees, costs-per-feature or recurring fees.
Analog Telephone FXS Ports 2 ports (both with lifeline capability in case of power outage)
PSTN Line FXO Ports 4 ports
Network Interfaces Dual Gigabit RJ45 ports with integrated PoE Plus (IEEE 802.3at-2009)
NAT Router Yes (supports router mode and switch mode)
Peripheral Ports USB,
SD
LED Indicators Power/Ready,
Network,
PSTN Line,
USB,
SD
LCD Display 128x32 graphic LCD with DOWN & OK button
Voice-over-Packet Capabilities LEC with NLP Packetized Voice Protocol Unit,
128ms-tail-length carrier grade Line Echo Cancellation, Dynamic Jitter Buffer,
Modem detection & auto-switch to G.711
Voice and Fax Codecs G.711 A-law/U-law,
G.722,
G.723.1 5.3K/6.3K,
G.726,
G.729A/B,
iLBC,
GSM,
AAL2-G.726-32,
ADPCM; T.38
Video Codecs H.264,
H.263,
H263+
QoS Layer 3 QoS,
Layer 2 QoS
DTMF Methods In Audio,
RFC2833,
and SIP INFO
Provisioning Protocol & Plug-and-Play TFTP/HTTP/HTTPS,
auto-discovery & auto-provisioning of Grandstream IP endpoints via ZeroConfig (DHCP Option 66 multicast SIP SUBSCRIBE mDNS),
eventlist between local and remote trunk
Network Protocols TCP/UDP/IP,
RTP/RTCP,
ICMP,
ARP,
DNS,
DDNS,
DHCP,
NTP,
TFTP,
SSH,
HTTP/HTTPS,
PPPoE,
SIP (RFC3261),
STUN,
SRTP,
TLS,
LDAP
Disconnect Methods Call Progress Tone,
Polarity Reversal,
Hook Flash Timing,
Loop Current Disconnect,
Busy Tone
Media Encryption SRTP,
TLS,
HTTPS,
SSH
Caller ID Bellcore/Telcordia,
ETSI-FSK,
ETSI-DTMF,
SIN 227 – BT
Polarity Reversal/Wink Yes,
with enable/disable option upon call establishment and termination
Call Center Multiple configurable call queues,
automatic call distribution (ACD) based on agent skills/availability/busy level,
in-queue announcement
Customizable Auto Attendant Up to 5 layers of IVR (Interactive Voice Response)
Maximum Call Capacity -Registered SIP devices- supports up to 500 registered SIP devices/users,
-Concurrent SIP calls- Up to 75 (UCM6204),
or 66% performance if calls are SRTP encrypted
Conference Bridges Up to 3,
password-protected conference bridges allowing up to 25 simultaneous PSTN or IP participants
Call Features Call park,
call forward,
call transfer,
DND,
ring/hunt group,
paging/intercom etc.
Mounting Wall mount & Desktop
Power Output 12VDC,
1.5A; Input 100 ~ 240VAC,
50 ~ 60Hz
Weight Unit weight 0.51kg,
Package weight 0.94kg
Environmental Operating 32 ~ 104ºF / 0 ~ 40ºC,
10 ~ 90% (non-condensing); Storage 14 ~ 140ºF / -10 ~ 60ºC
Dimensions 226mm L x 155mm W x 34.5mm H
Certificates FCC Part 15 (CFR 47) Class B,
Part 68,
CE EN55022 Class B,
EN55024,
EN61000-3-2,
EN61000-3-3,
EN60950-1,
TBR21,
RoHS,
A-TICK AS/NZS CISPR 22 Class B,
AS/NZS CISPR 24,
AS/NZS 60950,
AS/ACIF S002,
ITU-T K.21 (Basic Level),
UL 60950 (power adapter)
Manufacturer Grandstream
GRANDSTREAM
VoIP IP PBX, 4xFXO, 2xFXS (Grandstream UCM6204)
* For special orders only
Net Price: 478.00 USD  Unit: pcs  VAT: 23%

#05993

VoIP IP PBX, 8xFXO, 2xFXS (Grandstream UCM6208)

Designed to provide a centralized solution for the communication needs of businesses, the UCM6200 series IP PBX
appliance combines enterprise-grade voice, video, data, and mobility features in an easy-to-manage solution. This
IP PBX series allows businesses to unify multiple communication technologies, such as voice, video calling, video
conferencing, video surveillance, data tools, mobility options and facility access management onto one common
network that can be managed and/or accessed remotely. The secure and reliable UCM6200 series delivers enterprisegrade features without any licensing fees, costs-per-feature or recurring fees.
Analog Telephone FXS Ports 2 ports (both with lifeline capability in case of power outage)
PSTN Line FXO Ports 8 ports
Network Interfaces Dual Gigabit RJ45 ports with integrated PoE Plus (IEEE 802.3at-2009)
NAT Router Yes (supports router mode and switch mode)
Peripheral Ports USB,
SD
LED Indicators Power/Ready,
Network,
PSTN Line,
USB,
SD
LCD Display 128x32 graphic LCD with DOWN & OK button
Voice-over-Packet Capabilities LEC with NLP Packetized Voice Protocol Unit,
128ms-tail-length carrier grade Line Echo Cancellation, Dynamic Jitter Buffer,
Modem detection & auto-switch to G.711
Voice and Fax Codecs G.711 A-law/U-law,
G.722,
G.723.1 5.3K/6.3K,
G.726,
G.729A/B,
iLBC,
GSM,
AAL2-G.726-32,
ADPCM; T.38
Video Codecs H.264,
H.263,
H263+
QoS Layer 3 QoS,
Layer 2 QoS
DTMF Methods In Audio,
RFC2833,
and SIP INFO
Provisioning Protocol & Plug-and-Play TFTP/HTTP/HTTPS,
auto-discovery & auto-provisioning of Grandstream IP endpoints via ZeroConfig (DHCP Option 66 multicast SIP SUBSCRIBE mDNS),
eventlist between local and remote trunk
Network Protocols TCP/UDP/IP,
RTP/RTCP,
ICMP,
ARP,
DNS,
DDNS,
DHCP,
NTP,
TFTP,
SSH,
HTTP/HTTPS,
PPPoE,
SIP (RFC3261),
STUN,
SRTP,
TLS,
LDAP
Disconnect Methods Call Progress Tone,
Polarity Reversal,
Hook Flash Timing,
Loop Current Disconnect,
Busy Tone
Media Encryption SRTP,
TLS,
HTTPS,
SSH
Caller ID Bellcore/Telcordia,
ETSI-FSK,
ETSI-DTMF,
SIN 227 – BT
Polarity Reversal/Wink Yes,
with enable/disable option upon call establishment and termination
Call Center Multiple configurable call queues,
automatic call distribution (ACD) based on agent skills/availability/busy level,
in-queue announcement
Customizable Auto Attendant Up to 5 layers of IVR (Interactive Voice Response)
Maximum Call Capacity -Registered SIP devices- supports up to 800 registered SIP devices/users,
-Concurrent SIP calls- Up to 100 (UCM6204),
or 66% performance if calls are SRTP encrypted
Conference Bridges Up to 6,
password-protected conference bridges allowing up to 32 simultaneous PSTN or IP participants
Call Features Call park,
call forward,
call transfer,
DND,
ring/hunt group,
paging/intercom etc.
Mounting Rack mount & Desktop
Power Output 12VDC,
1.5A; Input 100 ~ 240VAC,
50 ~ 60Hz
Weight Unit weight 2.23kg,
Package weight 3.09kg
Environmental Operating 32 ~ 104ºF / 0 ~ 40ºC,
10 ~ 90% (non-condensing); Storage 14 ~ 140ºF / -10 ~ 60ºC
Dimensions 440mm L x 185mm W x 44mm H
Certificates FCC Part 15 (CFR 47) Class B,
Part 68,
CE EN55022 Class B,
EN55024,
EN61000-3-2,
EN61000-3-3,
EN60950-1,
TBR21,
RoHS,
A-TICK AS/NZS CISPR 22 Class B,
AS/NZS CISPR 24,
AS/NZS 60950,
AS/ACIF S002,
ITU-T K.21 (Basic Level),
UL 60950 (power adapter)
Manufacturer Grandstream
GRANDSTREAM
VoIP IP PBX, 8xFXO, 2xFXS (Grandstream UCM6208)
* For special orders only
Net Price: 799.00 USD  Unit: pcs  VAT: 23%

#05995

Automated failover solution for the UCM6510 (Grandstream HA100)

The HA100 offers an automated failover solution for the UCM6510 IP PBX. When connecting between
two UCM6510, the HA100 constantly monitors the operation status of both UCM6510 and automatically
switches the system control (including all of the connected telecom lines, network links, auxiliary devices,
and all of the SIP endpoints previously registered on the primary UCM6510) to the hot-standby secondary
UCM6510 in the event that the primary UCM6510 fails. It can complete the entire system switch between 10
and 50 seconds depending on the number of registered SIP endpoints. Thanks to its smart monitoring and
automated failover capability, the HA100 is an ideal high-availability solution for the UCM6510 to ensure
maximum total system reliability and uptime.
Analog Telephone FXS Ports 2 ports
PSTN Line FXO Ports 2 ports
T1/E1 Interface 1 port
Network Interfaces 1 LAN/ 1WAN
RS-485 2 (1 for Primary UCM6510 and 1 for Secondary UCM6510)
Reset Switch Yes
Universal Power Supply DC Power Port
Manufacturer Grandstream
GRANDSTREAM
Automated failover solution for the UCM6510 (Grandstream HA100)
* For special orders only
Net Price: 287.00 USD  Unit: pcs  VAT: 23%

 VoIP GATEWAYS
#06021

VoIP gateway 1xE1/T1 (Grandstream GXW4501)

The GXW4500 series are E1/T1 Digital VoIP Gateways that allow digital PSTN and ISDN trunks to be integrated with VoIP networks. By connecting the GXW4500 series with a VoIP network and a traditional PBX or E1/ T1 provider, businesses can drastically increase the amount of PSTN/ISDN trunks integrated with their VoIP network. The GXW4500 series offers three models that provide 1, 2 or 4 E1/T1/J1 spans and support 30, 60 or 120 concurrent calls to cater to the VoIP needs of large and medium sized enterprises.
E1/T1/J1 Interface 1 RJ45 ports,
supporting up to 30 simultaneous VoIP calls
Network protocols TCP/UDP/IP,
RTP/RTCP,
ICMP,
ARP,
DNS,
DDNS,
DHCP,
NTP,
TFTP,
SSH,
HTTP/HTTPS,
PPPoE,
STUN, SRTP,
TLS,
LDAP,
PPP,
Frame Relay (pending),
IPv6,
OpenVPN
Codecs G.711 A-law/U-law,
G.722,
G.723.1 5.3K/6.3K,
G.726,
G.729A/B,
iLBC,
AAL2-G.726-32
Ports 2x 10/100/1000 Mbps RJ-45,
2x USB 3.0,
1x SD card interface
LCD Display 128x32 dot matrix graphic LCD with DOWN and OK buttons
Fax over IP T.38 compliant Group 3 Fax Relay up to 14.4kpbs and auto-switch to G.711 for Fax Passthrough,
Fax data pump V.17,
V.21,
V.27ter,
V.29 for T.38 fax relay
QoS features Layer 2 QoS (802.1Q,
802.1p) and Layer 3 (ToS,
DiffServ,
MPLS) QoS
DTMF In-band audio,
RFC2833,
and SIP INFO
Device Management Syslog,
HTTPS,
Web browser,
voice prompt,
TR-069 management,
backup and restore,
port capture and packet capture
Power Input- 100 ~ 240VAC,
50/60Hz; Output- DC+12V,
2A
Operating temperatures 0C 45C
Operating humidity 10% 90%,
non-condensing
Dimensions 485mm(L) x 191mm(W) x 46.2mm (H)
Mounting Rack mount & Desktop
Certificates CE,
FCC
Warranty 24 months
Manufacturer Grandstream
GRANDSTREAM
VoIP gateway 1xE1/T1 (Grandstream GXW4501)
* For special orders only
Net Price: 719.00 USD  Unit: pcs  VAT: 23%

#06023

VoIP gateway 2xE1/T1 (Grandstream GXW4502)

The GXW4500 series are E1/T1 Digital VoIP Gateways that allow digital PSTN and ISDN trunks to be integrated with VoIP networks. By connecting the GXW4500 series with a VoIP network and a traditional PBX or E1/ T1 provider, businesses can drastically increase the amount of PSTN/ISDN trunks integrated with their VoIP network. The GXW4500 series offers three models that provide 1, 2 or 4 E1/T1/J1 spans and support 30, 60 or 120 concurrent calls to cater to the VoIP needs of large and medium sized enterprises.
E1/T1/J1 Interface 2 RJ45 ports,
supporting up to 60 simultaneous VoIP calls
Network protocols TCP/UDP/IP,
RTP/RTCP,
ICMP,
ARP,
DNS,
DDNS,
DHCP,
NTP,
TFTP,
SSH,
HTTP/HTTPS,
PPPoE,
STUN, SRTP,
TLS,
LDAP,
PPP,
Frame Relay (pending),
IPv6,
OpenVPN
Codecs G.711 A-law/U-law,
G.722,
G.723.1 5.3K/6.3K,
G.726,
G.729A/B,
iLBC,
AAL2-G.726-32
Ports 2x 10/100/1000 Mbps RJ-45,
2x USB 3.0,
1x SD card interface
LCD Display 128x32 dot matrix graphic LCD with DOWN and OK buttons
Fax over IP T.38 compliant Group 3 Fax Relay up to 14.4kpbs and auto-switch to G.711 for Fax Passthrough,
Fax data pump V.17,
V.21,
V.27ter,
V.29 for T.38 fax relay
QoS features Layer 2 QoS (802.1Q,
802.1p) and Layer 3 (ToS,
DiffServ,
MPLS) QoS
DTMF In-band audio,
RFC2833,
and SIP INFO
Device Management Syslog,
HTTPS,
Web browser,
voice prompt,
TR-069 management,
backup and restore,
port capture and packet capture
Power Input- 100 ~ 240VAC,
50/60Hz; Output- DC+12V,
2A
Operating temperatures 0C 45C
Operating humidity 10% 90%,
non-condensing
Dimensions 485mm(L) x 191mm(W) x 46.2mm (H)
Mounting Rack mount & Desktop
Certificates CE,
FCC
Warranty 24 months
Manufacturer Grandstream
GRANDSTREAM
VoIP gateway 2xE1/T1 (Grandstream GXW4502)
* For special orders only
Net Price: 1 210.00 USD  Unit: pcs  VAT: 23%

#05759
SELL OUT

VoIP gateway, 4xFXO, trunking (Edge-corE VG3306C)

Protocos SIP (RFC3261)
Voice codec G.711,
G.723,
G.729A/B
Ports 1x 10/100 Mbps RJ-45 (LAN),
1x 10/100 Mbps RJ-45 (WAN),
1x RJ-45 (console),
4x FXO,
1x RJ-45 (CDR)
FXO interface 2 wires,
Loop Start
Operational modes SIP Proxy client,
Peer-to-Peer
Tone DTMF relay
Fax support T38
Voice quality VAD,
CNG,
G.165/G.168 (echo cancelation) 16 ms,
adaptative Jitter buffer
Gain control +/- 6 dB
Call features auto attendant,
network operator group,
barring class,
trunk group,
trunk class,
line group,
call pickup,
network extension line,
inbound transit call,
outbound transit call,
Least Call Routing,
traffic management,
call transfer,
call forward,
abbreviated dialing,
CDR collection,
DND
Work with IP-PBX yes
STUN support yes
Biling system cooperation CDR records on separate port
Addressing DHCP,
static IP
Management console,
WWW,
phone set
Dimensions 172x117x35 mm
Operational temperature 0C 50C
Power 5V DC 1,5A; ~230V AC 50Hz
Warranty period 1 year
Manufacturer Edge-corE
EDGE-CORE
VoIP gateway,  4xFXO, trunking (Edge-corE VG3306C)
Net Price: 120.00 USD  Unit: pcs  VAT: 23%

#05978

VoIP gateway 4xFXO (Grandstream GXW4104)

Protocols SIP (RFC3261),
T38
Codecs G.711,
G.723,
G.729A/B,
GSM,
G.726
Ports 2x 10/100 Mbps RJ-45,
4x RJ-11 FXO,
Video IN
VoIP lines 3
Voice quality G.168 (echo cancelation),
dynamic jitter buffer
Fax support T38,
group 3 fax relay,
auto-switch to G.711 for Fax Pass-through
Video surveillance H.264, 30fps @ 352x240
Caller ID Bellcore typ 1 i 2,
ETSI,
BT,
NTT,
DTMF
QoS features DiffServ,
ToS,
802.1p
DTMF inband,
out of band,
SIP Info
Provisioning TFTP,
HTTP
PSTN signaling FXO Loop start
Addressing DHCP client,
DHCP server
Management WWW,
HTTPS
Power 12V DC,
~230V AC 50Hz
Operating temperatures 0C 40C
Operating humidity 10% 90%,
non-condensing
Dimensions 225x135x35 mm
Certificates CE,
FCC
Warranty 24 months
Manufacturer Grandstream
GRANDSTREAM
VoIP gateway  4xFXO (Grandstream GXW4104)
* For special orders only
Net Price: 257.00 USD  Unit: pcs  VAT: 23%

#05760
SELL OUT

VoIP gateway, 4xFXS (Edge-corE VG3306)

Protocos SIP (RFC3261)
Voice codec G.711,
G.723,
G.729A/B
Ports 1x 10/100 Mbps RJ-45 (LAN),
1x 10/100 Mbps RJ-45 (WAN),
1x RJ-45 (console),
4x FXS,
1x RJ-45 (CDR)
FXS interface 2 wires,
Loop Start
FXS feeding voltage 20 V
FXS feefing current 30 mA
FXO interface 2 wires,
Loop Start
Operational modes SIP Proxy client,
Peer-to-Peer
Tone DTMF relay
Fax support T38
Voice quality VAD,
CNG,
G.165/G.168 (echo cancelation) 16 ms,
adaptative Jitter buffer
Gain control +/- 6 dB
Call features speed dial,
SIP call forward,
inbound transit call,
outbound transit call,
call park,
E.164 dial plan,
authentication,
Hunting Group
Work with IP-PBX yes
STUN support yes
Biling system cooperation CDR records on separate port
Addressing DHCP,
static IP
Management console,
WWW,
phone set
Dimensions 172x117x35 mm
Operational temperature 0C 50C
Power 5V DC 1,5A; ~230V AC 50Hz
Warranty period 1 year
Manufacturer Edge-corE
EDGE-CORE
VoIP gateway,  4xFXS (Edge-corE VG3306)
Net Price: 145.00 USD  Unit: pcs  VAT: 23%

#06024

VoIP gateway 4xE1/T1 (Grandstream GXW4504)

The GXW4500 series are E1/T1 Digital VoIP Gateways that allow digital PSTN and ISDN trunks to be integrated with VoIP networks. By connecting the GXW4500 series with a VoIP network and a traditional PBX or E1/ T1 provider, businesses can drastically increase the amount of PSTN/ISDN trunks integrated with their VoIP network. The GXW4500 series offers three models that provide 1, 2 or 4 E1/T1/J1 spans and support 30, 60 or 120 concurrent calls to cater to the VoIP needs of large and medium sized enterprises.
E1/T1/J1 Interface 4 RJ45 ports,
supporting up to 120 simultaneous VoIP calls
Network protocols TCP/UDP/IP,
RTP/RTCP,
ICMP,
ARP,
DNS,
DDNS,
DHCP,
NTP,
TFTP,
SSH,
HTTP/HTTPS,
PPPoE,
STUN, SRTP,
TLS,
LDAP,
PPP,
Frame Relay (pending),
IPv6,
OpenVPN
Codecs G.711 A-law/U-law,
G.722,
G.723.1 5.3K/6.3K,
G.726,
G.729A/B,
iLBC,
AAL2-G.726-32
Ports 2x 10/100/1000 Mbps RJ-45,
2x USB 3.0,
1x SD card interface
LCD Display 128x32 dot matrix graphic LCD with DOWN and OK buttons
Fax over IP T.38 compliant Group 3 Fax Relay up to 14.4kpbs and auto-switch to G.711 for Fax Passthrough,
Fax data pump V.17,
V.21,
V.27ter,
V.29 for T.38 fax relay
QoS features Layer 2 QoS (802.1Q,
802.1p) and Layer 3 (ToS,
DiffServ,
MPLS) QoS
DTMF In-band audio,
RFC2833,
and SIP INFO
Device Management Syslog,
HTTPS,
Web browser,
voice prompt,
TR-069 management,
backup and restore,
port capture and packet capture
Power Input- 100 ~ 240VAC,
50/60Hz; Output- DC+12V,
2A
Operating temperatures 0C 45C
Operating humidity 10% 90%,
non-condensing
Dimensions 485mm(L) x 191mm(W) x 46.2mm (H)
Mounting Rack mount & Desktop
Certificates CE,
FCC
Warranty 24 months
Manufacturer Grandstream
GRANDSTREAM
VoIP gateway 4xE1/T1 (Grandstream GXW4504)
* For special orders only
Net Price: 2 020.00 USD  Unit: pcs  VAT: 23%

#05979

VoIP gateway 8xFXO (Grandstream GXW4108)

Protocols SIP (RFC3261),
T38
Codecs G.711,
G.723,
G.729A/B,
GSM,
G.726
Ports 2x 10/100 Mbps RJ-45,
8x RJ-11 FXO,
Video IN
VoIP lines 3
Voice quality G.168 (echo cancelation),
dynamic jitter buffer
Fax support T38,
group 3 fax relay,
auto-switch to G.711 for Fax Pass-through
Video surveillance H.264, 30fps @ 352x240
Caller ID Bellcore typ 1 i 2,
ETSI,
BT,
NTT,
DTMF
QoS features DiffServ,
ToS,
802.1p
DTMF inband,
out of band,
SIP Info
Provisioning TFTP,
HTTP
PSTN signaling FXO Loop start
Addressing DHCP client,
DHCP server
Management WWW,
HTTPS
Power 12V DC,
~230V AC 50Hz
Operating temperatures 0C 40C
Operating humidity 10% 90%,
non-condensing
Dimensions 225x135x35 mm
Certificates CE,
FCC
Warranty 24 months
Manufacturer Grandstream
GRANDSTREAM
VoIP gateway  8xFXO (Grandstream GXW4108)
Net Price: 346.00 USD  Unit: pcs  VAT: 23%

#05981

VoIP gateway, 8xFXS (Grandstream HT818)

Protocols SIP (RFC3261)
Voice codec G.711 with Annex I (PLC) and Annex II (VAD/CNG),
G.723.1,
G.729A/B,
G.726,
iLBC,
OPUS,
dynamic jitter buffer,
advanced line echo cancellation
Ports 1x 10/100/1000 Mbps RJ-45 (WAN),
1x 10/100/1000 Mbps RJ-45 (LAN)
Voice ports 8xFXS
Operating modes SIP Proxy client
DTMF In-audio,
RFC2833 and/or SIP INFO
VoIP lines 8
Fax support T.38 compliant Group 3 Fax Relay up to 14.4kpbs and auto-switch to G.711 for Fax Pass-through
QoS Layer 2 (802.1Q VLAN,
SIP/RTP 802.1p) and Layer 3 (ToS,
Diffserv,
MPLS)
Voice quality VAD,
CNG,
Packet Loss Concacelation (PLC),
dynamic Jitter buffer,
echo cancelation
Call features Caller ID display or block,
call waiting,
flash,
blind or attended transfer,
forward,
hold,
do not disturb,
3-way conference
Router with NAT function
Addressing static IP,
DHCP client
Provisioning HTTP,
HTTPS,
SSH,
TFTP,
TR-069 ,
secure and automated provisioning using AES encryption,
syslog
Management WWW,
telnet
Power 12V DC 1,5A,
100-240VAC,
50-60Hz
Operating temperatures 0C 40C
Operating humidity 10% 90%,
non-condensing
Dimensions 180x120x36 mm
Certificates CE,
FCC,
RCM
Warranty 24 months
Manufacturer Grandstream
GRANDSTREAM
VoIP gateway, 8xFXS (Grandstream HT818)
Net Price: 175.00 USD  Unit: pcs  VAT: 23%

#06025

VoIP gateway 16xFXS (Grandstream GXW4216)

The GXW4200 high-density FXS gateway series enables businesses of all sizes to create an easy-to-deploy VoIP solution that takes advantage of Gigabit speeds. These FXS gateways offer the ability to seamlessly connect multiple locations and all devices within an office to any hosted or on premise IP PBX network to make deployments as easy as possible. The GXW4200 series includes 16/24/32/48 FXS ports and a Gigabit network port. Deploy the GXW4200 series to allow any businesses to create a cost-effective hybrid IP and analog telephone system that allows them to enjoy the benefits of VoIP communications while preserving investment on existing analog phones, Fax machines and legacy PBX systems.
Protocols SIP (RFC3261),
T38
Codecs G.711,
G.723,
G.726,
G.729A/B/E,
iLBC
Ports 2x 10/100/1000 Mbps RJ-45,
16x RJ-11 FXS,
1x 50-pin Telco connector
VoIP lines 2
Voice quality G.168 (echo cancelation),
dynamic jitter buffer
Life Line yes
Caller ID Bellcore type 1 & 2,
ETSI,
BT,
NTT,
DTMF
Fax support T38 group 3 fax relay,
auto switch to G.711 for Fax Pass-through
QoS functions DiffServ,
ToS,
802.1P/Q
DTMF inband,
out of band,
SIP Info
Provisioning TFTP,
HTTP,
HTTPS
Security SIPS,
TLS
Addressing DHCP client,
DHCP server
Management WWW,
console,
telnet,
HTTPS
Power 12V DC,
~230V AC 50Hz
Operating temperatures 0C 40C
Operating humidity 10% 90%,
non-condensing
Dimensions 440mm (L) x 185mm (W) x 44mm (H) (1U)
Certificates FCC Part 15 (CFR 47) Class B,
CE EN55022 Class B,
EN55024,
EN61000-3-2,
EN16000-3-3,
EN60950-1,
RoHS,
C-TICK AS/NZS,
CISPR 22 Class B,
AS/NZS CISPR 24,
AN/NZS 60950,
ITU-T K.21 (Basic Test Level),
UL 60950 (power adapter)
Warranty 24 months
Manufacturer Grandstream
GRANDSTREAM
VoIP gateway 16xFXS (Grandstream GXW4216)
* For special orders only
Net Price: 444.00 USD  Unit: pcs  VAT: 23%

#06480

VoIP gateway 24xFXS (Grandstream GXW4224)

The GXW4200 high-density FXS gateway series enables businesses of all sizes to create an easy-to-deploy VoIP solution that takes advantage of Gigabit speeds. These FXS gateways offer the ability to seamlessly connect multiple locations and all devices within an office to any hosted or on premise IP PBX network to make deployments as easy as possible. The GXW4200 series includes 16/24/32/48 FXS ports and a Gigabit network port. Deploy the GXW4200 series to allow any businesses to create a cost-effective hybrid IP and analog telephone system that allows them to enjoy the benefits of VoIP communications while preserving investment on existing analog phones, Fax machines and legacy PBX systems.
Protocols SIP (RFC3261),
T38
Codecs G.711,
G.723,
G.726,
G.729A/B/E,
iLBC
Ports 2x 10/100/1000 Mbps RJ-45,
24x RJ-11 FXS,
1x 50-pin Telco connector
VoIP lines 2
Voice quality G.168 (echo cancelation),
dynamic jitter buffer
Life Line yes
Caller ID Bellcore type 1 & 2,
ETSI,
BT,
NTT,
DTMF
Fax support T38 group 3 fax relay,
auto switch to G.711 for Fax Pass-through
QoS functions DiffServ,
ToS,
802.1P/Q
DTMF inband,
out of band,
SIP Info
Provisioning TFTP,
HTTP,
HTTPS
Security SIPS,
TLS
Addressing DHCP client,
DHCP server
Management WWW,
console,
telnet,
HTTPS
Power 12V DC,
~230V AC 50Hz
Operating temperatures 0C 40C
Operating humidity 10% 90%,
non-condensing
Dimensions 440mm (L) x 185mm (W) x 44mm (H) (1U)
Certificates FCC Part 15 (CFR 47) Class B,
CE EN55022 Class B,
EN55024,
EN61000-3-2,
EN16000-3-3,
EN60950-1,
RoHS,
C-TICK AS/NZS,
CISPR 22 Class B,
AS/NZS CISPR 24,
AN/NZS 60950,
ITU-T K.21 (Basic Test Level),
UL 60950 (power adapter)
Warranty 24 months
Manufacturer Grandstream
GRANDSTREAM
VoIP gateway 24xFXS (Grandstream GXW4224)
Net Price: 639.00 USD  Unit: pcs  VAT: 23%

#05988

VoIP gateway 32xFXS (Grandstream GXW4232)

The GXW4200 high-density FXS gateway series enables businesses of all sizes to create an easy-to-deploy VoIP solution that takes advantage of Gigabit speeds. These FXS gateways offer the ability to seamlessly connect multiple locations and all devices within an office to any hosted or on premise IP PBX network to make deployments as easy as possible. The GXW4200 series includes 16/24/32/48 FXS ports and a Gigabit network port. Deploy the GXW4200 series to allow any businesses to create a cost-effective hybrid IP and analog telephone system that allows them to enjoy the benefits of VoIP communications while preserving investment on existing analog phones, Fax machines and legacy PBX systems.
Protocols SIP (RFC3261),
T38
Codecs G.711,
G.723,
G.726,
G.729A/B/E,
iLBC
Ports 2x 10/100/1000 Mbps RJ-45,
32x RJ-11 FXS,
1x 50-pin Telco connector
VoIP lines 2
Voice quality G.168 (echo cancelation),
dynamic jitter buffer
Life Line yes
Caller ID Bellcore type 1 & 2,
ETSI,
BT,
NTT,
DTMF
Fax support T38 group 3 fax relay,
auto switch to G.711 for Fax Pass-through
QoS functions DiffServ,
ToS,
802.1P/Q
DTMF inband,
out of band,
SIP Info
Provisioning TFTP,
HTTP,
HTTPS
Security SIPS,
TLS
Addressing DHCP client,
DHCP server
Management WWW,
console,
telnet,
HTTPS
Power 12V DC,
~230V AC 50Hz
Operating temperatures 0C 40C
Operating humidity 10% 90%,
non-condensing
Dimensions 440mm (L) x 185mm (W) x 44mm (H) (1U)
Certificates FCC Part 15 (CFR 47) Class B,
CE EN55022 Class B,
EN55024,
EN61000-3-2,
EN16000-3-3,
EN60950-1,
RoHS,
C-TICK AS/NZS,
CISPR 22 Class B,
AS/NZS CISPR 24,
AN/NZS 60950,
ITU-T K.21 (Basic Test Level),
UL 60950 (power adapter)
Warranty 24 months
Manufacturer Grandstream
GRANDSTREAM
VoIP gateway 32xFXS (Grandstream GXW4232)
* For special orders only
Net Price: 758.00 USD  Unit: pcs  VAT: 23%

#06027

VoIP gateway 48xFXS (Grandstream GXW4248)

The GXW4200 high-density FXS gateway series enables businesses of all sizes to create an easy-to-deploy VoIP solution that takes advantage of Gigabit speeds. These FXS gateways offer the ability to seamlessly connect multiple locations and all devices within an office to any hosted or on premise IP PBX network to make deployments as easy as possible. The GXW4200 series includes 16/24/32/48 FXS ports and a Gigabit network port. Deploy the GXW4200 series to allow any businesses to create a cost-effective hybrid IP and analog telephone system that allows them to enjoy the benefits of VoIP communications while preserving investment on existing analog phones, Fax machines and legacy PBX systems.
Protocols SIP (RFC3261),
T38
Codecs G.711,
G.723,
G.726,
G.729A/B/E,
iLBC
Ports 2x 10/100/1000 Mbps RJ-45,
48x RJ-11 FXS,
1x 50-pin Telco connector
VoIP lines 2
Voice quality G.168 (echo cancelation),
dynamic jitter buffer
Life Line yes
Caller ID Bellcore type 1 & 2,
ETSI,
BT,
NTT,
DTMF
Fax support T38 group 3 fax relay,
auto switch to G.711 for Fax Pass-through
QoS functions DiffServ,
ToS,
802.1P/Q
DTMF inband,
out of band,
SIP Info
Provisioning TFTP,
HTTP,
HTTPS
Security SIPS,
TLS
Addressing DHCP client,
DHCP server
Management WWW,
console,
telnet,
HTTPS
Power 12V DC,
~230V AC 50Hz
Operating temperatures 0C 40C
Operating humidity 10% 90%,
non-condensing
Dimensions 440mm (L) x 255mm (W) x 44mm (H) (1U)
Certificates FCC Part 15 (CFR 47) Class B,
CE EN55022 Class B,
EN55024,
EN61000-3-2,
EN16000-3-3,
EN60950-1,
RoHS,
C-TICK AS/NZS,
CISPR 22 Class B,
AS/NZS CISPR 24,
AN/NZS 60950,
ITU-T K.21 (Basic Test Level),
UL 60950 (power adapter)
Warranty 24 months
Manufacturer Grandstream
GRANDSTREAM
VoIP gateway 48xFXS (Grandstream GXW4248)
* For special orders only
Net Price: 1 110.00 USD  Unit: pcs  VAT: 23%

 VoIP ATA
#05982

VoIP gateway, 1xFXS (Grandstream HT801)

The HT801 is a single port analog telephone adapter (ATA) that allows users to create a high-quality and
manageable IP telephony solution for residential and office environments. Its ultra-compact size, voice
quality, advanced VoIP functionality, security protection and auto provisioning options enable users to take
advantage of VoIP on analog phones. It also allows service providers to offer high quality IP service to their
market. The HT801 is an ideal ATA for individual use as well as commercial IP voice deployments worldwide.
Network Interface One 10M/100Mbps auto-sensing Ethernet port(RJ45)
FXS Port 1
Voicemail Indicator Yes
Fax over IP T.38 Compliant Group 3 Fax Relay up to 14.4kbps and auto switch to G.711 for Fax Pass-through
Caller ID Bellcore Type 1 & 2,
ETSI,
BT,
NTT,
and DTMF-based CID
Remote Configuration HTTP/HTTPS/Telnet/TFTP Provisioning
Security Media SRTP
Warranty 24 months
GRANDSTREAM
VoIP gateway, 1xFXS (Grandstream HT801)
Net Price: 45.10 USD  Unit: pcs  VAT: 23%

#05985

VoIP gateway, 1xFXS, 1xFXO (Grandstream HT813)

The HT813 is an analog telephone adapter that features 1 analog telephone FXS port and 1 PSTN line FXO
port in order to offer backup lifeline support using a PSTN line. The integration of a FXO and FXS port
enables this hybrid ATA to support remote calling to and from the PSTN line. For added flexibility, the FXS
port extends VoIP service to one analog device. Users can convert their analog technology to VoIP thanks to
the HT813’s ultra-compact size, HD voice quality, advanced VoIP functionality, high-end security protection
and multiple auto provisioning options. These advanced features also allow service providers to offer high
quality IP service to customers looking to upgrade to VoIP.
Network Interface Two 10M/100Mbps auto-sensing Ethernet port(RJ45)
FXS Port 1
FXO Port 1
Voicemail Indicator Yes
Fax over IP T.38 Compliant Group 3 Fax Relay up to 14.4kbps and auto switch to G.711 for Fax Pass-through
Caller ID Bellcore Type 1 & 2,
ETSI,
BT,
NTT,
and DTMF-based CID
Remote Configuration HTTP/HTTPS/Telnet/TFTP Provisioning
Security Media SRTP
Warranty 24 months
GRANDSTREAM
VoIP gateway, 1xFXS, 1xFXO (Grandstream HT813)
* For special orders only
Net Price: 87.30 USD  Unit: pcs  VAT: 23%

#06312

VoIP gateway, 2xFXS (Grandstream HT802)

Network Interface One 10M/100Mbps auto-sensing Ethernet port(RJ45)
FXS Port 2
Voicemail Indicator Yes
Telephony Features Caller ID display or block,
call waiting,
flash,
blind or attended transfer,
forward,
hold, do not disturb, 3-way conference
Caller ID Bellcore typ 1 & 2,
ETSI,
BT,
NTT i CID based on DTMF
Remote Configuration HTTP,
HTTPS,
SSH,
TFTP,
TR-069 ,
secure and automated provisioning using AES encryption,
syslog
Security Media SRTP
Warranty 24 months
GRANDSTREAM
VoIP gateway, 2xFXS (Grandstream HT802)
Net Price: 51.20 USD  Unit: pcs  VAT: 23%

#06311

VoIP gateway, 2xFXS (Grandstream HT812)

The HT812 is an advanced 2-port analog telephone adapter (ATA) with 2 FXS ports and an integrated Gigabit
NAT router. Built upon Grandstream’s market-leading SIP ATA/gateway technology with millions of units
successfully deployed worldwide, this powerful ATA features exceptional voice quality in various application
environments, strong encryption with unique security certificate per unit, automated provisioning for
volume deployment and device management, and outstanding network performance for home and office
use.
Network Interface Two 10M/100/1000Mbps auto-sensing Ethernet port(RJ45)
FXS Port 2
Voicemail Indicator Yes
Telephony Features Caller ID display or block,
call waiting,
flash,
blind or attended transfer,
forward,
hold, do not disturb, 3-way conference
Caller ID Bellcore typ 1 & 2,
ETSI,
BT,
NTT i CID based on DTMF
Remote Configuration HTTP,
HTTPS,
SSH,
TFTP,
TR-069 ,
secure and automated provisioning using AES encryption,
syslog
Security Media SRTP
Warranty 24 months
GRANDSTREAM
VoIP gateway, 2xFXS (Grandstream HT812)
Net Price: 60.20 USD  Unit: pcs  VAT: 23%

#05986

VoIP gateway, 4xFXS (Grandstream HT814)

The HT814 is an advanced 4-port VoIP gateway with 4 FXS ports and an integrated Gigabit NAT router.
Built upon Grandstream’s market-leading SIP ATA/gateway technology with millions of units successfully
deployed worldwide, this powerful gateway features exceptional voice quality in various application
environments, strong encryption with unique security certificate per unit, automated provisioning for
volume deployment and device management, and outstanding network performance for home and office
use.
Network Interface Two 10M/100/1000Mbps auto-sensing Ethernet port(RJ45)
FXS Port 4
Voicemail Indicator Yes
Fax over IP T.38 Compliant Group 3 Fax Relay up to 14.4kbps and auto switch to G.711 for Fax Pass-through
Caller ID Bellcore Type 1 & 2,
ETSI,
BT,
NTT,
and DTMF-based CID
Remote Configuration HTTP/HTTPS/Telnet/TFTP Provisioning
Security Media SRTP
Warranty 24 months
GRANDSTREAM
VoIP gateway, 4xFXS (Grandstream HT814)
Net Price: 113.00 USD  Unit: pcs  VAT: 23%

 VoIP PHONES
#06698

DECT Cordless HD Handset for Mobility (Grandstream DP720)

The DP720 is a DECT cordless VoIP phone that allows users to mobilize their VoIP network throughout any business, warehouse, retail store and residential environment. It is supported by Grandstream’s DP750 DECT VoIP base station (#06697) and delivers a combination of mobility and top-notch telephony performance. Up to five DP720 handsets are supported on each DP750 while each DP720 supports a range of up to 300 meters outdoors and 50 meters indoors from the base station. The DP720 touts a suite of top-notch telephony features including support for up to 10 SIP accounts per handset, full HD audio, a 3.5mm headset jack, multi-language support, a speakerphone and more. When paired with Grandstream’s DP750 DECT Base
Station, the DP720 offers a powerful DECT VoIP handset that allows any business or residential user to create a cordless VoIP solution.
Protocols Hearing Aid Compatibility (HAC) compliant
Telephony standards DECT
Frequency bands 1880 – 1900 MHz (Europe),
1920 – 1930 MHz (US),
1910 – 1920 MHz (Brazil),
1786 – 1792 MHz (Korea),
1893 – 1906 MHz (Japan),
1880 – 1895 MHz (Taiwan)
Number of Channels 10 (Europe),
5 (US,
Brazil or Japan),
3 (Korea),
8 (Taiwan)
Range up to 300 meters outdoors and 50 meters indoors
Peripherals 1.8 inch (128x160) color TFT LCD,
23 keys including 2 soft keys,
5 navigation/ menu keys,
4 dedicated function keys for SEND,
POWER/END,
SPEAKERPHONE,
MUTE,
3-color MWI LED,
3.5mm headset jack,
Removable belt clip,
Micro-USB port for alternative charging and non-battery operation
Voice codec G.722 codec for HD audio and G.726 codec for narrow band audio (G.711μ/a-law,
G.723.1,
G.729A/B,
iLBC and OPUS are supported via companion DECT base station DP750),
AEC,
AGC,
Ambient noise reduction
Call features Hold,
transfer,
forward,
3-way conference,
call park,
call pickup,
downloadable phonebook,
call waiting,
call log,
auto answer,
click-to-dial,
flexible dial plan,
music on hold
HD Audio Yes,
both on Handset and Speakerphone
Security DECT authentication & encryption
Multi-language Chinese Simple,
Chinese Tradition,
Czech,
Danish,
Dutch,
English,
Estonian,
Finnish,
French,
German,
Hebrew,
Hungarian,
Japanese,
Korean,
Norwegian,
Polish,
Portuguese,
Romanian,
Spanish,
Turkish
Multi-line Access Each handset may access up to ten (10) lines
Power & Green Energy Efficiency Universal Power Supply Input AC 100-240V 50/60Hz,
Output 5VDC 1A,
Micro-USB connection,
Rechargeable 800mAh Ni-MH Low Self-Discharge (LSD) AAA batteries (250 hours of standby time and 20 hours of talk time)
Package Content Handset unit,
universal power supply,
charger cradle,
belt clip,
2 batteries,
Quick Start Guide
Dimensions (H x W x D) Handset- 155 x 50 x 26 mm,
charger cradle- 35 x 63.5 x 54 mm
Weight Handset 138g,
Charger Cradle- 71g; Universal Power Supply- 50g,
Package- 360g
Temperature and Humidity Operation -10º to 50ºC (14 to 122ºF),
Charging- 0 to 45ºC (32 to 113ºF)
Storage -20º to 60ºC (-4 to 140ºF),
Humidity 10% to 90% non-condensing
Compliance FCC Part 15D,
47 CFR 2.1093 & IEEE1528-2013,
Part 68,
Part 15B,
CE EN60950,
EN301489-1-6,
EN301406,
EN50360,
EN62209-1,
RCM AS/NZS60950,
AS/ACIF S004,
ANATEL- 2288-16-9452
GRANDSTREAM
DECT Cordless HD Handset for Mobility (Grandstream DP720)
Net Price: 57.20 USD  Unit: pcs  VAT: 23%

#06015

DECT Cordless HD Handset for Mobility (Grandstream DP722)

The DP722 is a mid-tier DECT cordless IP phone that allows users to mobilize their VoIP network throughout any business, warehouse, retail store and residential environment. It is supported by Grandstream’s DP750 and DP752 DECT VoIP base stations and delivers a combination of mobility and excellent telephony performance. Up to five DP722 handsets to be supported on each base station while each DP722 supports a range of up to 350 meters outdoors (with DP752) and 50 meters indoors, 20 hours of talk time and 250-hour standby time. It touts a suite of robust features including support for up to 10 SIP accounts per handset, full HD audio, 1.8 inch color display, a 3.5mm headset jack, push-to-talk, a speakerphone and more. When paired with Grandstream’s DECT Base Stations, the DP722 offers an affordable mid-range cordless DECT solution for any business or residential user.
Protocols Hearing Aid Compatibility (HAC) compliant
Telephony standards DECT
Frequency bands 1880 – 1900 MHz (Europe),
1920 – 1930 MHz (US),
1910 – 1920 MHz (Brazil),
1786 – 1792 MHz (Korea),
1893 – 1906 MHz (Japan),
1880 – 1895 MHz (Taiwan)
Number of Channels 10 (Europe),
5 (US,
Brazil or Japan),
3 (Korea),
8 (Taiwan)
Range up to 300 meters outdoors and 50 meters indoors
Peripherals 1.8 inch (128x160) color TFT LCD,
23 keys including 2 soft keys,
5 navigation/ menu keys,
4 dedicated function keys for SEND,
POWER/END,
SPEAKERPHONE,
MUTE,
3-color MWI LED,
3.5mm headset jack,
Removable belt clip,
Micro-USB port for alternative charging and non-battery operation
Voice codec G.722 codec for HD audio and G.726 codec for narrow band audio (G.711μ/a-law,
G.723.1,
G.729A/B,
iLBC and OPUS are supported via companion DECT base station DP750),
AEC,
AGC,
Ambient noise reduction,
advanced noise suppression for incoming audio
Call features Hold,
transfer,
forward,
3-way conference,
call park,
call pickup,
downloadable phonebook,
call waiting,
call log,
auto answer,
click-to-dial,
flexible dial plan,
music on hold
HD Audio Yes,
both on Handset and Speakerphone
Security DECT authentication & encryption
Multi-language Chinese Simple,
Chinese Tradition,
Czech,
Danish,
Dutch,
English,
Estonian,
Finnish,
French,
German,
Hebrew,
Hungarian,
Japanese,
Korean,
Norwegian,
Polish,
Portuguese,
Romanian,
Spanish,
Turkish
Multi-line Access Each handset may access up to ten (10) lines
Power & Green Energy Efficiency Universal Power Supply Input AC 100-240V 50/60Hz,
Output 5VDC 1A,
Micro-USB connection,
Rechargeable 800mAh Ni-MH Low Self-Discharge (LSD) AAA batteries (250 hours of standby time and 20 hours of talk time)
Package Content Handset unit,
universal power supply,
charger cradle,
belt clip,
2 batteries,
Quick Start Guide
Dimensions (H x W x D) Handset- 158 x 50 x 28.1mm,
charger cradle- 81.15 x 75.89 x 36.36mm
Weight Handset Handset 110g,
Charger cradle 44g,
Universal power supply 50g,
Package 328g
Temperature and Humidity Operation -10º to 50ºC (14 to 122ºF),
Charging- 0 to 45ºC (32 to 113ºF)
Storage -20º to 60ºC (-4 to 140ºF),
Humidity 10% to 90% non-condensing
Compliance FCC- FCC Part 15B; FCC Part 15D; SAR (FCC 47 CFR Part2.1093; IEEE 1528; IEC 62209-2); FCC Part68 HAC; FCC ID,
CE- EN 55032; EN 55035; EN 61000-3-2; EN 61000-3-3; EN 60950-1;EN 301 489-1/-6; EN 301 406; EN 50332-2; SAR(EN50360;EN50566;EN 50663;EN62209-1; EN62209-2; EN 62479); RED NB Cert,
RCM- AS/NZS CISPR32; AS/NZS 60950.1;AS/CA S004;AS/ACIF S040. ANATEL,
EAC,
UL (adapter)
GRANDSTREAM
DECT Cordless HD Handset for Mobility (Grandstream DP722)
Net Price: 60.20 USD  Unit: pcs  VAT: 23%

#06016

DECT Cordless HD Handset for Mobility (Grandstream DP730)

The DP730 is a DECT cordless IP phone that allows users to mobilize their VoIP network throughout any business, warehouse, retail store and residential environment. It is supported by Grandstream’s DP750 and DP752 DECT VoIP base stations and delivers a combination of mobility and top-notch telephony performance. Up to five DP730 handsets are supported on each base station while each DP730 supports a range of up to 400 meters outdoors (with DP752) and 50 meters indoors along with 40 hours of talk time and 500-hour standby time. It touts a suite of robust telephony features including support for up to 10 SIP accounts per handset, full HD audio, 2.4 inch color display, a 3.5mm headset jack, push-to-talk, a speakerphone and more. When paired with Grandstream’s DECT Base Stations, the DP730 is a high-end handset that offers a powerful cordless DECT solution for any business or residential user.
Protocols Hearing Aid Compatibility (HAC) compliant
Telephony standards DECT
Frequency bands 1880 – 1900 MHz (Europe),
1920 – 1930 MHz (US),
1910 – 1920 MHz (Brazil),
1786 – 1792 MHz (Korea),
1893 – 1906 MHz (Japan),
1880 – 1895 MHz (Taiwan)
Number of Channels 10 (Europe),
5 (US,
Brazil or Japan),
3 (Korea),
8 (Taiwan)
Range up to 400 meters (DP752) or up to 300 meters (DP750) and 50 meters indoors
Peripherals 2.4 inch (240x320) color TFT LCD,
27 keys including 3 soft keys,
5 navigation/ menu keys,
4 dedicated function keys for SEND,
POWER/END,
SPEAKERPHONE,
MUTE,
3-color MWI LED,
3.5mm headset jack,
Removable belt clip,
Micro-USB port for alternative charging and non-battery operation
Voice codec G.722 codec for HD audio and G.726 codec for narrow band audio (G.711μ/a-law,
G.723.1,
G.729A/B,
iLBC and OPUS are supported via companion DECT base station DP750),
AEC,
AGC,
Ambient noise reduction,
advanced noise suppression for incoming audio
Call features Hold,
transfer,
forward,
3-way conference,
call park,
call pickup,
downloadable phonebook,
call waiting,
call log,
auto answer,
click-to-dial,
flexible dial plan,
music on hold
HD Audio Yes,
both on Handset and Speakerphone
Security DECT authentication & encryption
Multi-language Chinese Simple,
Chinese Tradition,
Czech,
Danish,
Dutch,
English,
Estonian,
Finnish,
French,
German,
Hebrew,
Hungarian,
Japanese,
Korean,
Norwegian,
Polish,
Portuguese,
Romanian,
Spanish,
Turkish
Multi-line Access Each handset may access up to ten (10) lines
Power & Green Energy Efficiency Universal Power Supply Input AC 100-240V 50/60Hz,
Output 5VDC 1A,
Micro-USB connection,
Rechargeable 1500mAh Ni-MH Low Self-Discharge (LSD) AAA batteries (250 hours of standby time and 20 hours of talk time)
Package Content Handset unit,
universal power supply,
charger cradle,
belt clip,
2 batteries,
Quick Start Guide
Dimensions (H x W x D) Handset- 168.5 x 52.5 x 21.8mm,
charger cradle- 76 x 73 x 81mm
Weight Handset Handset 180g,
Charger cradle 78g,
Universal power supply 50g,
Package 465g
Temperature and Humidity Operation -10º to 50ºC (14 to 122ºF),
Charging- 0 to 45ºC (32 to 113ºF)
Storage -20º to 60ºC (-4 to 140ºF),
Humidity 10% to 90% non-condensing
Compliance FCC- FCC Part 15B; FCC Part 15D; SAR (FCC 47 CFR Part2.1093; IEEE 1528; IEC 62209-2); FCC Part68 HAC; FCC ID,
CE- EN 55032; EN 55035; EN 61000-3-2; EN 61000-3-3; EN 60950-1;EN 301 489-1/-6; EN 301 406; EN 50332-2; SAR(EN50360;EN50566;EN 50663;EN62209-1; EN62209-2; EN 62479); RED NB Cert,
RCM- AS/NZS CISPR32; AS/NZS 60950.1;AS/CA S004;AS/ACIF S040. ANATEL,
EAC,
UL (adapter)
GRANDSTREAM
DECT Cordless HD Handset for Mobility (Grandstream DP730)
Net Price: 99.30 USD  Unit: pcs  VAT: 23%

#06012

Extension module (Grandstream KGXP2200EXT)

The GXP2200 Extension module is the ideal choice for providing powerful call control and flexibility to any user. The module itself boasts an intuitive and clear design, with 20 dual-colored extension keys and 2 arrow keys for page switching. Each module supports visibility for up to 40 additional contacts and extensions, while also supporting the ability to connect up to 4 GXP2200EXT modules to compatible Grandstream phones for visibility on up to 160 new contacts/extensions. Users can enjoy the added productivity that the GXP2200EXT brings with its feature rich design. It supports traditional call features on each of its programmable buttons, including bridged line appearance/shared call appearance, busy lamp fields, call park/pick-up, speed dial, presence, intercom, and conference/transfer/forward.
Supported by the GXP2140, GXP2170 and GXV3240.
GRANDSTREAM
Extension module (Grandstream KGXP2200EXT)
* For special orders only
Net Price: 114.00 USD  Unit: pcs  VAT: 23%

#06697

Long-range DECT VoIP Base Station (Grandstream DP750)

The DP750 is a powerful DECT VoIP base station that pairs with up to 5 of Grandstream’s DP720 DECT handsets (#06698) to offer mobility to business and residential users. It supports a range of 300 meters outdoors and 50 meters indoors to give users the freedom to move around their work or home space, delivering efficient flexibility. This DECT VoIP base station supports up to 10 SIP accounts and 5 concurrent calls while also offering 3-way voice conferencing, full HD audio and integrated PoE. A shared SIP account on all handsets will add seamless unified features that gives users the ability to answer all calls regardless of location in real-time. The DP750 supports a variety of auto-provisioning methods and TLS/SRTP/HTTPS encryption security. When paired with Grandstream’s DP720, the DP750 offers a powerful DECT VoIP base station that allows any business or residential user to create a cordless VoIP solution.
Protocols SIP RFC3261,
TCP/IP/UDP,
RTP/RTCP,
HTTP/HTTPS,
ARP/RARP,
ICMP,
DNS (A record,
SRV,
NAPTR),
DHCP,
PPPoE,
SSH,
TFTP,
NTP,
STUN,
SIMPLE,
LLDP-MED,
LDAP,
TR-069, 802.1x,
TLS,
SRTP,
IPv6 (Pending)
Ports 1* 10/100 Mbps auto-sensing Ethernet port with integrated PoE
Telephony standards DECT
Frequency bands 1880 – 1900 MHz (Europe),
1920 – 1930 MHz (US),
1910 – 1920 MHz (Brazil),
1786 – 1792 MHz (Korea),
1893 – 1906 MHz (Japan),
1880 – 1895 MHz (Taiwan)
Number of Channels 10 (Europe),
5 (US,
Brazil or Japan),
3 (Korea),
8 (Taiwan)
Range up to 300 meters outdoors and 50 meters indoors
Peripherals 5 LED indicators- Power,
Network,
Register,
Call,
DECT Reset button,
Pairing/Paging button
Voice codec G.711μ/a-law,
G.723.1,
G.729A/B,
G.726-32,
iLBC,
G.722,
OPUS,
G.722.2/AMR-WB (special order),
in-band and out-of-band DTMF (in audio,
RFC2833,
SIP INFO),
VAD,
CNG,
PLC,
AJB
Call features Hold,
transfer,
forward,
3-way conference,
downloadable phonebook (XML,
LDAP,
up to 3000 entries),
call waiting,
call log (up to 300 records),
auto answer,
flexible dial plan,
music on hold,
server redundancy and fail-over
QoS Layer 2 QoS (802.1Q,
802.1P) and Layer 3 QoS (ToS,
DiffServ,
MPLS)
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,
DECT authentication & encryption
Multi-language Chinese Simple,
Chinese Tradition,
Czech,
Danish,
Dutch,
English,
Estonian,
Finnish,
French,
German,
Hebrew,
Hungarian,
Japanese,
Korean,
Norwegian,
Polish,
Portuguese,
Romanian,
Spanish,
Turkish
Multi-line Access Each handset may access up to ten (10) lines
Ring Group Flexible options when multiple handsets share the same SIP account,
Circular Mode- all phones ring sequentially from the phone next to the one that,
answered last,
Linear Mode- all phones ring sequentially in the predesignated order,
Parallel Mode- all phones ring concurrently and after one phone answers,
the remaining available phones can make new calls,
Shared Mode-all phones ring concurrently and always share the same line similar to analog phones
Power & Green Energy Efficiency Universal Power Supply Input AC 100-240V 50/60Hz,
Output 5VDC 1A,
Micro-USB connection,
PoE- IEEE802.3af Class 1, 0.44W–3.84W
Package Content Base Unit,
Universal Power Supply,
Ethernet cable,
Quick Start Guide,
GPL statement
Dimensions (H x W x D) 28.5 x 130 x 90 mm
Weight Handset Base unit- 143g,
Universal Power Supply- 50g,
Package- 360g
Temperature and Humidity Operation -10º to 50ºC (14 to 122ºF)
Storage -20º to 60ºC (-4 to 140ºF),
Humidity 10% to 90% non-condensing
Compliance FCC- Part 15D,
Part 15B,
CE- EN60950,
EN301489-1-6,
EN301406,
RCM- AS/NZS60950,
ANATEL- 2288-16-9452
GRANDSTREAM
Long-range DECT VoIP Base Station (Grandstream DP750)
Net Price: 54.20 USD  Unit: pcs  VAT: 23%

#06014

Long-range DECT VoIP Base Station (Grandstream DP752)

The DP752 is a powerful DECT VoIP base station that pairs with up to 5 of Grandstream’s DP series DECT handsets to offer mobility to business and residential users. It supports outdoor range of up to 400 meters with the DP730 or up to 350 meters with DP722/DP720 as well as indoor range up to 50 meters to give users the freedom to move around their work or home. This DECT VoIP base station supports up to 10 SIP accounts and 5 concurrent calls while also offering 3-way voice conferencing, full HD audio and integrated PoE. A shared SIP account on all handsets will add seamless unifed features that gives users the ability to answer all calls regardless of location in real-time. The DP752 supports a variety of auto-provisioning methods and TLS/SRTP/HTTPS encryption security. When paired with Grandstream’s DP720, DP722 or DP730 handsets, the DP752 offers a powerful cordless DECT solution for any business or residential user.
Protocols SIP RFC3261,
TCP/IP/UDP,
RTP/RTCP,
HTTP/HTTPS,
ARP/RARP,
ICMP,
DNS (A record,
SRV,
NAPTR),
DHCP,
PPPoE,
SSH,
TFTP,
NTP,
STUN,
SIMPLE,
LLDP-MED,
LDAP,
TR-069, 802.1x,
TLS,
SRTP,
IPv6 (Pending)
Ports 1* 10/100 Mbps auto-sensing Ethernet port with integrated PoE
Telephony standards DECT
Frequency bands 1880 – 1900 MHz (Europe),
1920 – 1930 MHz (US),
1910 – 1920 MHz (Brazil),
1786 – 1792 MHz (Korea),
1893 – 1906 MHz (Japan),
1880 – 1895 MHz (Taiwan)
Number of Channels 10 (Europe),
5 (US,
Brazil or Japan),
3 (Korea),
8 (Taiwan)
Range up to 400 meters (DP730) or up to 350 meters (DP722/DP720) and 50 meters indoors
Peripherals 3 LED indicators- Power,
Network,
DECT
Voice codec G.711μ/a-law,
G.723.1,
G.729A/B,
G.726-32,
iLBC,
G.722,
OPUS,
G.722.2/AMR-WB (special order),
in-band and out-of-band DTMF (in audio,
RFC2833,
SIP INFO),
VAD,
CNG,
PLC,
AJB
Call features Hold,
transfer,
forward,
3-way conference,
downloadable phonebook (XML,
LDAP,
up to 3000 entries),
call waiting,
call log (up to 300 records),
auto answer,
flexible dial plan,
music on hold,
server redundancy and fail-over
QoS Layer 2 QoS (802.1Q,
802.1P) and Layer 3 QoS (ToS,
DiffServ,
MPLS)
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,
DECT authentication & encryption
Multi-language Chinese Simple,
Chinese Tradition,
Czech,
Danish,
Dutch,
English,
Estonian,
Finnish,
French,
German,
Hebrew,
Hungarian,
Japanese,
Korean,
Norwegian,
Polish,
Portuguese,
Romanian,
Spanish,
Turkish
Multi-line Access Each handset may access up to ten (10) lines
Ring Group Flexible options when multiple handsets share the same SIP account,
Circular Mode- all phones ring sequentially from the phone next to the one that,
answered last,
Linear Mode- all phones ring sequentially in the predesignated order,
Parallel Mode- all phones ring concurrently and after one phone answers,
the remaining available phones can make new calls,
Shared Mode-all phones ring concurrently and always share the same line similar to analog phones
Power & Green Energy Efficiency Universal Power Supply Input AC 100-240V 50/60Hz,
Output 5VDC 1A,
Micro-USB connection,
PoE- IEEE802.3af Class 1, 0.44W–3.84W
Package Content Base Unit,
Universal Power Supply,
Ethernet cable,
Quick Start Guide,
GPL statement
Dimensions (H x W x D) 140.31 x 64.98 x 105mm
Weight Handset Base unit 140g,
Universal power supply 50g,
Package 370g
Temperature and Humidity Operation -10º to 50ºC (14 to 122ºF)
Storage -20º to 60ºC (-4 to 140ºF),
Humidity 10% to 90% non-condensing
Compliance FCC- FCC Part 15B; FCC Part 15D; MPE; FCC ID,
CE- EN 55032; EN 55035; EN 61000-3-2; EN 61000-3-3; EN 60950-1; EN 301 489-1/-6; EN 301 406; EN 50385; RED NB Cert,
RCM- AS/NZS 32; AS/NZS 60950.1,
ANATEL- ANATEL,
EAC,
UL (adapter)
GRANDSTREAM
Long-range DECT VoIP Base Station (Grandstream DP752)
Net Price: 54.20 USD  Unit: pcs  VAT: 23%

#06696

Long-range DECT VoIP repeater (Grandstream DP760)

The DP760 is a powerful wideband DECT repeater (wireless relay station) that auto associates to Grandstream’s DP750 DECT base station offering extended mobility to business and residential users. The DP760 extends an additional range of 300 meters outdoors and 50 meters indoors to give users the freedom to move around their home or work space. This Wideband DECT Repeater relays up to 2 concurrent HD calls. The Ethernet connection provides PoE for convenient installation and a variety of remote features including provisioning, status monitoring and repeater firmware upgrades. When paired with Grandstream’s DP750 DECT VoIP base station and DP720 handsets, the DP760 offers a powerful extended DECT solution for users looking to add coverage to their VoIP DECT system.
Protocols TCP/IP/UDP,
HTTP/HTTPS,
ARP/RARP,
ICMP,
DNS,
DHCP,
PPPoE,
SSH,
TFTP,
NTP,
LLDP-MED,
UPnP
Ports 1* 10/100 Mbps auto-sensing Ethernet port with integrated PoE
Telephony standards DECT EN 301 406-2001,
DECT GAP TBR22 EN 300 444-2001,
DECT WRS EN 300 700,
CAT-iq TS 102 527
Frequency bands 1880 – 1900 MHz (Europe),
1920 – 1930 MHz (US),
1910 – 1920 MHz (Brazil)
Number of Channels 10 (Europe),
5 (US,
Brazil)
Range up to 300 meters outdoors and 50 meters indoors
Peripherals 5 LED indicators- Power,
Network,
Association,
Activity,
DECT Signal Strength,
Reset button,
Dissociation button
Voice codec G.722 codec for HD audio and G.726 codec for narrow band audio
Call features Plug-n-Play,
auto association,
auto region detection and seamless call handover
Security User and administrator level access control,
MD5 and MD5-sess based authentication,
256-bit AES encrypted configuration file,
HTTPS,
802.1x media access control
Multi-language Arabic,
Chinese Simple,
Chinese Tradition,
Czech,
Dutch,
English,
French,
German,
Hebrew,
Italian,
Japanese,
Korean,
Polish,
Portuguese,
Russian,
Serbian,
Slovakian,
Spanish,
Swedish,
Turkish
Association Up to 5 repeaters in star,
Relays up to 2 concurrent HD calls,
Automatic or manual association to base station
Ring Group Flexible options when multiple handsets share the same SIP account,
Circular Mode- all phones ring sequentially from the phone next to the one that,
answered last,
Linear Mode- all phones ring sequentially in the predesignated order,
Parallel Mode- all phones ring concurrently and after one phone answers,
the remaining available phones can make new calls,
Shared Mode-all phones ring concurrently and always share the same line similar to analog phones
Power & Green Energy Efficiency Universal Power Supply Input AC 100-240V 50/60Hz,
Output 5VDC 1A,
Micro-USB connection,
PoE- IEEE802.3af Class 1, 0.44W–3.84W
Package Content Base Unit,
Universal Power Supply,
Ethernet cable,
Quick Start Guide,
GPL statement
Dimensions (H x W x D) 28.5 x 130 x 90 mm
Weight Handset Base unit- 143g,
Universal Power Supply- 50g,
Package- 360g
Temperature and Humidity Operation -10º to 50ºC (14 to 122ºF)
Storage -20º to 60ºC (-4 to 140ºF),
Humidity 10% to 90% non-condensing
Compliance FCC- Part 15D,
Part 15B,
MPE,
CE- EN60950,
EN301489-1-6,
EN301406,
RCM- AS/NZS60950,
ANATEL- 2288-16-9452
GRANDSTREAM
Long-range DECT VoIP repeater (Grandstream DP760)
* For special orders only
Net Price: 119.00 USD  Unit: pcs  VAT: 23%

#06017

Portable WiFi phone (Grandstream WP820)

The WP820 is a portable WiFi phone designed to suit a variety of enterprises and vertical market applications, including retail, logistics, medical and security. This powerful, portable WiFi phone comes equipped with integrated dual-band 802.11a/b/g/n WiFi support, advanced antenna design and roaming support, and integrated Bluetooth for pairing with headsets and mobile devices. By adding 7.5 hour talk time and HD voice with dual-MICs, the WP820 offers a powerful combination of features, mobility and durability to suit all portable telephony needs. The GMC08, a battery charging pack for the WP820 that can charge up to 8 batteries at a time, is available seperately.
Protocols SIP RFC3261,
TCP/IP/UDP,
RTP/RTCP,
HTTP/HTTPS,
ARP,
ICMP,
DNS (A record,
SRV,
NAPTR),
DHCP,
SSH,
TFTP,
NTP,
STUN,
SIMPLE,
LLDP-MED,
LDAP,
TR-069, 802.1x,
TLS,
SRTP,
IPv6
Bluetooth Yes,
Bluetooth 4.2
Operating System Android 7.0,
supports custom Android apps that fit the phone’s screen/keyboard
Peripherals 2.4 inch (240x320) color TFT LCD,
27 keys including 3 soft keys,
5 navigation/ menu keys,
4 dedicated function keys for SEND,
POWER/END,
SPEAKERPHONE,
MUTE,
3-color MWI LED,
3.5mm headset jack,
Removable belt clip,
Micro-USB port for alternative charging and non-battery operation
Voice codec G.722 codec for HD audio and G.726 codec for narrow band audio (G.711μ/a-law,
G.723.1,
G.729A/B,
iLBC and OPUS are supported via companion DECT base station DP750),
AEC,
AGC,
Ambient noise reduction,
advanced noise suppression for incoming audio
Call features Hold,
transfer,
forward,
3-way conference,
call park,
call pickup,
downloadable phonebook,
call waiting,
call log,
auto answer,
click-to-dial,
flexible dial plan,
music on hold
HD Audio Yes,
both on handset and speakerphone with support for wideband audio,
HAC supported
Security User and administrator level passwords,
MD5 and MD5-sess based authentication,
256-bit AES based secure configuration file,
SRTP,
TLS,
802.1x media access control
Multi-language English,
Arabic,
Chinese,
Czech,
Dutch,
German,
French,
Hebrew,
Italian,
Japanese,
Polish,
Portuguese,
Russian,
Spanish,
Turkish and more
Power & Green Energy Efficiency Universal Power Supply Input AC 100-240V 50/60Hz,
Output 5VDC 1A (5W),
Micro-USB connection,
Rechargeable 1500mAh Ni-MH Low Self-Discharge (LSD) AAA batteries (150 hours of standby time and 7,5 hours of talk time)
Dimensions (H x W x D) Handset- 168.5 x 52.5 x 21.8mm,
charger cradle- 76 x 73 x 81mm
Weight Handset Handset 161g,
Package 456g
Temperature and Humidity Operation 0º to 45ºC
Storage -20º to 60ºC (-4 to 140ºF),
Humidity 10% to 90% non-condensing
Compliance FCC,
CE,
RCM,
EAC
GRANDSTREAM
Portable WiFi phone (Grandstream WP820)
Net Price: 214.00 USD  Unit: pcs  VAT: 23%

#06020

Video Phone with Android (Grandstream GXV3370)

The GXV3370 is a powerful desktop video phone for enterprise users. It features a 7" touch screen, advanced megapixel camera for HD video conferencing, built-in WiFi and Bluetooth, Gigabit network speeds and innovative telephony functionalities. It also runs on Android 7.0 and has flexible SDK support for custom apps. The GXV3370 is fully interoperable with nearly all major SIP platforms on the market and can be seamlessly integrated with Grandstream’s portfolio including SIP based security cameras, door systems, IP PBXs, and video conferencing systems and services. This video phone is the perfect choice for users looking for an integrated video communications solution for their desktop.
Protocols/Standards SIP RFC3261,
TCP/IP/UDP,
RTP/RTCP,
HTTP/HTTPS,
ARP,
ICMP,
DNS (rekord A,
SRV,
NAPTR),
DHCP,
PPPoE,
SSH,
TFTP,
NTP,
STUN,
SIMPLE,
LLDP-MED,
LDAP,
TR-069, 802.1x,
TLS,
SRTP,
IPv6,
OpenVPN
Networking Interfaces Dual-switched 10/10/10000Mbps with PoE/PoE+
WiFi Yes,
dual-band 802.11a/b/g/n (2.4GHz & 5GHz)
Bluetooth Yes,
Bluetooth 4.0 + EDR
Graphic Display 7" 1024600 capacitive touch screen (5 points) TFT LCD
Camera Tiltable mega pixel CMOS camera with privacy shutter,
720p 30fps
Auxiliary Ports RJ9 headset jack (allowing EHS with Plantronics headsets),
USB,
SD,
HDMI-out (1.4 up to 720p 30fps)
Feature Keys 2 function touch keys VOLUME +/-,
3 dedicated Android touch keys HOME,
MENU,
and BACK
Voice Codec Support f or G.711µ/a,
G.722 (wide-band),
G.726-32,
iLBC,
Opus,
G.729 A/B,
in-band and outof-band DTMF (In audio,
RFC2833,
SIP INFO),
CNG,
PLC,
AGC,
AJB
Telephony Features Hold,
transfer,
forward (unconditional/no-answer/busy),
call park/pickup,
6-way audio conference,
shared-call-appearance (SCA) / bridged-line-appearance (BLA),
virtual MPK,
downloadable phone book (XML,
LDAP),
call waiting,
call history,
boss-secretary virtual button,
flexible dial plan,
hot desking,
personalized music ringtones,
server redundancy & fail-over
Sample Applications Skype,
Google Hangouts,
Microsoft Lync,
Web browser,
Adobe Flash,
Facebook,
Twitter,
YouTube,
news,
weather,
stock,
Internet radio,
Pandora,
Last.fm,
Yahoo Flickr,
Photobucket,
alarm clock,
Google calendar,
mobile phone data import/export via Bluetooth,
etc. API/SDK available for advanced custom application development
HD Audio Yes,
HD handset with support for wideband audio
Base Stand Yes,
1 angle position available
QoS Layer 2 QoS (802.1Q,
802.1p) and Layer 3 (ToS,
DiffServ,
MPLS) QoS
Security User and administrator level passwords,
MD5 and MD5-sess based authentication,
256-bit AES encrypted configuration file,
TLS,
SRTP,
HTTPS,
802.1x media access control
Multi-language English,
German,
Italian,
French,
Spanish,
Portuguese,
Russian,
Croatian,
Chinese,
Korean,
Japanese,
and more
Upgrade/Provisioning Firmware upgrade via TFTP / HTTP / HTTPS or local HTTP upload,
mass provisioning using TR069 or AES encrypted XML configuration file
Power & Green Energy Efficiency Universal power adapter included- Input 100-240VAC 50-60Hz; Output 12VDC,
1.5A (18W),
Integrated PoE+ (Power-over-Ethernet)* 802.3at,
Class 4
Temperature and Humidity Operation 0C to 40C,
Storage -10C to 60C,
Humidity 10% to 90% Non-condensing
Compliance FCC- Part 15 (CFR 47) Class B; UL 60950 (power adapter); Part 68 (HAC),
CE- EN55022 Class B,
EN55024,
EN61000-3-2,
EN61000-3-3,
EN60950-1,
EN62479,
RoHS,
RCM- AS/ACIF S004; AS/NZS CISPR22/24; AS/NZS 60950; AS/NZS 4268
GRANDSTREAM
Video Phone with Android (Grandstream GXV3370)
* For special orders only
Net Price: 343.00 USD  Unit: pcs  VAT: 23%

#06033

VoIP phone (Grandstream GRP2602)

Part of the GRP series of Carrier-Grade IP Phones, the GRP2602 is an essential 2-line model designed with zerotouch provisioning for mass deployment and easy management. It features a sleek design and a suite of nextgeneration features including Wi-Fi support (GRP2602W), 5-way voice conferencing to maximize productivity, integrated PoE (GRP2602P), full HD audio on both the speaker and handset to allow users to communicate with the utmost clarity, EHS support for Plantronics, Jabra, and Sennheiser headsets and multi-language support.
Protocols/Standards 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
Networking Interfaces Dual-switched 10/100Mbps port
Feature Keys 2 line keys with dual-color LED and support for 4 SIP account,
4 XML programmable context sensitive soft keys,
5 (navigation,
menu) keys. 8 dedicated function keys for- MESSAGE(with LED indicator),
TRANSFER, HEADSET,
MUTE,
SEND/REDIAL,
SPEAKERPHONE,
VOL+,
VOL
Telephony Features Hold,
transfer,
forward,
3-way conference (via programmable key),
call waiting,
off-hook auto dial,
click-to-dial,
flexible dial plan,
personalized music ringtones,
server redundancy and fail-over
HD Audio Yes,
HD handset with support for wideband audio
Base Stand Yes,
2 angle position available
GRANDSTREAM
Grandstream GRP2602, VoIP phone
Net Price: 79.70 USD  Unit: pcs  VAT: 23%

#06040
NEW

VoIP phone (Grandstream GRP2604)

Part of the GRP series of Carrier-Grade IP Phones, the GRP2604 is an essential 3-line model designed with zerotouch provisioning for mass deployment and easy management. It features a sleek design and a suite of nextgeneration features including: 5-way voice conferencing to maximize productivity, integrated PoE (GRP2604P),
full HD audio on both the speaker and handset to allow users to communicate with the utmost clarity, EHS
support for Plantronics, Jabra, and Sennheiser headsets and multi-language support. The GRP series includes
carrier-grade security features to provide enterprise-level security, including secure boot, dual firmware images
and encrypted data storage. For cloud provisioning and centralized management, the GRP2604 is supported
by Grandstream’s Device Management System (GDMS), which provides a centralized interface to configure,
provision, manage and monitor deployments of Grandstream endpoints. Built for the needs of on-site or
remote desktop workers and designed for easy deployment by enterprises, service providers and other highvolume markets, the GRP2604 offers an easy-to-use and easy-to-deploy voice endpoint.
Protocols/Standards 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
Networking Interfaces Dual-switched 10/100Mbps port
Feature Keys 2 line keys with dual-color LED and support for 4 SIP account,
4 XML programmable context sensitive soft keys,
5 (navigation,
menu) keys. 8 dedicated function keys for- MESSAGE(with LED indicator),
TRANSFER, HEADSET,
MUTE,
SEND/REDIAL,
SPEAKERPHONE,
VOL+,
VOL
Telephony Features Hold,
transfer,
forward,
3-way conference (via programmable key),
call waiting,
off-hook auto dial,
click-to-dial,
flexible dial plan,
personalized music ringtones,
server redundancy and fail-over
HD Audio Yes,
HD handset with support for wideband audio
Base Stand Yes,
2 angle position available
GRANDSTREAM
VoIP phone (Grandstream GRP2604)
Net Price: 63.00 USD  Unit: pcs  VAT: 23%

#06039
NEW

VoIP phone (Grandstream GRP2604P)

Part of the GRP series of Carrier-Grade IP Phones, the GRP2604 is an essential 3-line model designed with zerotouch provisioning for mass deployment and easy management. It features a sleek design and a suite of nextgeneration features including: 5-way voice conferencing to maximize productivity, integrated PoE (GRP2604P),
full HD audio on both the speaker and handset to allow users to communicate with the utmost clarity, EHS
support for Plantronics, Jabra, and Sennheiser headsets and multi-language support. The GRP series includes
carrier-grade security features to provide enterprise-level security, including secure boot, dual firmware images
and encrypted data storage. For cloud provisioning and centralized management, the GRP2604 is supported
by Grandstream’s Device Management System (GDMS), which provides a centralized interface to configure,
provision, manage and monitor deployments of Grandstream endpoints. Built for the needs of on-site or
remote desktop workers and designed for easy deployment by enterprises, service providers and other highvolume markets, the GRP2604 offers an easy-to-use and easy-to-deploy voice endpoint.
Protocols/Standards 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
Networking Interfaces Dual-switched 10/100Mbps port,
PoE
Feature Keys 2 line keys with dual-color LED and support for 4 SIP account,
4 XML programmable context sensitive soft keys,
5 (navigation,
menu) keys. 8 dedicated function keys for- MESSAGE(with LED indicator),
TRANSFER, HEADSET,
MUTE,
SEND/REDIAL,
SPEAKERPHONE,
VOL+,
VOL
Telephony Features Hold,
transfer,
forward,
3-way conference (via programmable key),
call waiting,
off-hook auto dial,
click-to-dial,
flexible dial plan,
personalized music ringtones,
server redundancy and fail-over
HD Audio Yes,
HD handset with support for wideband audio
Base Stand Yes,
2 angle position available
GRANDSTREAM
VoIP phone (Grandstream GRP2604P)
Net Price: 63.00 USD  Unit: pcs  VAT: 23%

#06008

VoIP phone (Grandstream GRP2612)

The GRP2612 is a powerful 2-line carrier-grade IP phone designed with zero-touch provisioning for mass deployment and easy management. Built for the needs of desktop workers and designed for easy deployment by enterprises, service providers and other high-volume markets, the GRP2612 offers an easy-to-use and easy-to deploy voice endpoint.
Protocols/Standards 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
Networking Interfaces Dual-switched 10/100Mbps port
Feature Keys 2 SIP account,
4 line keys,
3-way conferencing,
4 XML programmable context-sensitive soft keys
Voice Codec Support for G.723.1,
G.729A/B,
G.711µ/a,
G.726-32,
G.722 (wide-band),
iLBC,
in-band and out-of-band DTMF (in audio,
RFC2833,
SIP INFO
Telephony Features Hold,
transfer,
forward,
3-way conference (via programmable key),
call waiting,
off-hook auto dial,
click-to-dial,
flexible dial plan,
personalized music ringtones,
server redundancy and fail-over
HD Audio Yes,
HD handset with support for wideband audio
Base Stand Yes,
2 angle position available
GRANDSTREAM
VoIP phone (Grandstream GRP2612)
Net Price: 72.20 USD  Unit: pcs  VAT: 23%

#06009

VoIP phone (Grandstream GRP2612P)

The GRP2612 is a powerful 2-line carrier-grade IP phone designed with zero-touch provisioning for mass deployment and easy management. Built for the needs of desktop workers and designed for easy deployment by enterprises, service providers and other high-volume markets, the GRP2612 offers an easy-to-use and easy-to deploy voice endpoint.
Protocols/Standards 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
Networking Interfaces Dual-switched 10/100Mbps port PoE
Feature Keys 2 SIP account,
4 line keys,
3-way conferencing,
4 XML programmable context-sensitive soft keys
Voice Codec Support for G.723.1,
G.729A/B,
G.711µ/a,
G.726-32,
G.722 (wide-band),
iLBC,
in-band and out-of-band DTMF (in audio,
RFC2833,
SIP INFO
Telephony Features Hold,
transfer,
forward,
3-way conference (via programmable key),
call waiting,
off-hook auto dial,
click-to-dial,
flexible dial plan,
personalized music ringtones,
server redundancy and fail-over
HD Audio Yes,
HD handset with support for wideband audio
Base Stand Yes,
2 angle position available
GRANDSTREAM
VoIP phone (Grandstream GRP2612P)
Net Price: 72.20 USD  Unit: pcs  VAT: 23%

#06010

VoIP phone (Grandstream GRP2613)

The GRP2613 is a powerful 3-line carrier-grade IP phone designed with zero-touch provisioning for mass deployment and easy management. Built for the needs of desktop workers and designed for easy deployment by enterprises, service providers and other high-volume markets, the GRP2613 offers an easy-to-use and easy-to deploy voice endpoint.
Protocols/Standards 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
Networking Interfaces Dual-switched 10/100/1000Mbps port PoE
Feature Keys 3 SIP account,
6 line keys,
3-way conferencing,
4 XML programmable context-sensitive soft keys
Voice Codec Support for G.723.1,
G.729A/B,
G.711µ/a,
G.726-32,
G.722 (wide-band),
iLBC,
in-band and out-of-band DTMF (in audio,
RFC2833,
SIP INFO
Telephony Features Hold,
transfer,
forward,
3-way conference (via programmable key),
call waiting,
off-hook auto dial,
click-to-dial,
flexible dial plan,
personalized music ringtones,
server redundancy and fail-over
HD Audio Yes,
HD handset with support for wideband audio
Base Stand Yes,
2 angle position available
GRANDSTREAM
VoIP phone (Grandstream GRP2613)
* For special orders only
Net Price: 97.80 USD  Unit: pcs  VAT: 23%

#06011

VoIP phone (Grandstream GRP2614)

The GRP2614 features a sleek design and a suite of next-generation features including dual LCD screens with 40 virtual multi-purpose keys (VPKs), integrated WiFi, Bluetooth support, dual Gigabit ports and more. The GRP series includes carrier-grade security features to provide enterprise-level security, including secure boot, dual firmware images and encrypted data storage. For cloud provisioning and centralized management, the GRP2614 is supported by Grandstream’s Device Management System (GDMS), which provides a centralized interface to configure, provision, manage and monitor deployments of Grandstream endpoints.
Protocols/Standards 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
Networking Interfaces Dual-switched 10/100/1000Mbps port PoE
Bluetooth Yes,
integrated
Wi-Fi Yes,
integrated dual-band WiFi 802.11 a/b/g/n/ac (2.4Ghz & 5Ghz)
Feature Keys 4 SIP account,
4 line keys,
3-way conferencing,
4 XML programmable context-sensitive soft keys
Voice Codec Support for G.723.1,
G.729A/B,
G.711µ/a,
G.726-32,
G.722 (wide-band),
iLBC,
in-band and out-of-band DTMF (in audio,
RFC2833,
SIP INFO
Telephony Features Hold,
transfer,
forward,
3-way conference (via programmable key),
call waiting,
off-hook auto dial,
click-to-dial,
flexible dial plan,
personalized music ringtones,
server redundancy and fail-over
HD Audio Yes,
HD handset with support for wideband audio
Base Stand Yes,
2 angle position available
GRANDSTREAM
VoIP phone (Grandstream GRP2614)
* For special orders only
Net Price: 172.00 USD  Unit: pcs  VAT: 23%

#08964

VoIP phone (Grandstream GXP1610)

The GXP1610 is a simple IP phone for a small businesses (SMBs) or home office use. This Linux-based model features a single SIP account, up to 2 call appearances, and 3 XML programmable soft keys. A 132x48 LCD screen creates a clear display for easy viewing. Additional features such as dual switched 10/100 Mbps ports, multi-language support and 3-way conferencing allow the GXP1610 to be a high quality, user-friendly and dependable office phone.
Protocols/Standards SIP RFC3261,
TCP/IP/UDP,
RTP,
HTTP/HTTPS,
ARP/RARP,
ICMP,
DNS (A record,
SRV,
NAPTR),
DHCP,
PPPoE,
TELNET,
TFTP,
NTP,
STUN,
TR-069, 802.1x
Networking Interfaces Dual-switched 10/100Mbps port
Feature Keys 1 SIP account,
2 line keys,
3-way conferencing,
3 XML programmable context-sensitive soft keys
Voice Codec Support for G.723.1,
G.729A/B,
G.711µ/a,
G.726-32,
G.722 (wide-band),
iLBC,
in-band and out-of-band DTMF (in audio,
RFC2833,
SIP INFO
Telephony Features Hold,
transfer,
forward,
3-way conference (via programmable key),
call waiting,
off-hook auto dial,
click-to-dial,
flexible dial plan,
personalized music ringtones,
server redundancy and fail-over
HD Audio Yes,
HD handset with support for wideband audio
Base Stand Yes,
1 angle position available
GRANDSTREAM
VoIP phone (Grandstream GXP1610)
Net Price: 43.60 USD  Unit: pcs  VAT: 23%

#08965

VoIP phone (Grandstream GXP1625)

The GXP1620/1625 is Grandstream’s standard IP phone for small businesses. This Linux-based, 2-line IP Phone includes 3-way conferencing to keep workers in-touch and productive. A 132x48 backlit LCD screen creates a clear display for easy viewing. Additional features such as dual-switched 10/100mbps ports, HD audio, multi-language support, integrated PoE (GXP1625 only) and 3 XML programmable soft keys allow the GXP1620/1625 to be a high quality, versatile and dependable office phone.
Protocols/Standards SIP RFC3261,
TCP/IP/UDP,
RTP,
HTTP/HTTPS,
ARP/RARP,
ICMP,
DNS (A record,
SRV,
NAPTR),
DHCP,
PPPoE,
TELNET,
TFTP,
NTP,
STUN,
TR-069, 802.1x
Networking Interfaces Dual-switched 10/100Mbps port PoE
Feature Keys 2 SIP account,
2 line keys,
3-way conferencing,
3 XML programmable context-sensitive soft keys
Voice Codec Support for G.723.1,
G.729A/B,
G.711µ/a,
G.726-32,
G.722 (wide-band),
iLBC,
in-band and out-of-band DTMF (in audio,
RFC2833,
SIP INFO
Telephony Features Hold,
transfer,
forward,
3-way conference (via programmable key),
call waiting,
off-hook auto dial,
click-to-dial,
flexible dial plan,
personalized music ringtones,
server redundancy and fail-over
HD Audio Yes,
HD handset with support for wideband audio
Base Stand Yes,
1 angle position available
GRANDSTREAM
VoIP phone (Grandstream GXP1625)
Net Price: 55.70 USD  Unit: pcs  VAT: 23%

#08966

VoIP phone (Grandstream GXP1628)

The GXP1628 is a powerful Gigabit IP phone designed for small businesses. This Linux-based, 2-line IP Phone model includes 8 BLF keys and 3-way conferencing to keep workers in-touch and productive. A 132x48 backlit LCD screen creates a clear display for easy viewing. Additional features such as dual HD audio, multi-language support, integrated PoE and 3 XML programmable allow the GXP1628 to be a high quality, versatile and dependable office phone.
Protocols/Standards SIP RFC3261,
TCP/IP/UDP,
RTP,
HTTP/HTTPS,
ARP/RARP,
ICMP,
DNS (A record,
SRV,
NAPTR),
DHCP,
PPPoE,
TELNET,
TFTP,
NTP,
STUN,
TR-069, 802.1x
Networking Interfaces Dual-switched 10/100/1000Mbps port PoE
Feature Keys 2 SIP account,
2 line keys,
3-way conferencing,
3 XML programmable context-sensitive soft keys
Voice Codec Support for G.723.1,
G.729A/B,
G.711µ/a,
G.726-32,
G.722 (wide-band),
iLBC,
in-band and out-of-band DTMF (in audio,
RFC2833,
SIP INFO
Telephony Features Hold,
transfer,
forward,
3-way conference (via programmable key),
call waiting,
off-hook auto dial,
click-to-dial,
flexible dial plan,
personalized music ringtones,
server redundancy and fail-over
HD Audio Yes,
HD handset with support for wideband audio
Base Stand Yes,
1 angle position available
GRANDSTREAM
VoIP phone (Grandstream GXP1628)
Net Price: 73.70 USD  Unit: pcs  VAT: 23%

#08967

VoIP phone (Grandstream GXP1630)

The GXP1630 is a powerful Gigabit IP phone designed for small businesses. This Linux-based, 3-line IP Phone model includes 8 BLF keys and 4-way conferencing to keep workers in-touch and productive. A 132x64 backlit LCD screen creates a clear display for easy viewing. Additional features such as dual HD audio, multi-language support, integrated PoE and 3 XML programmable allow the GXP1630 to be a high quality, versatile and dependable office phone.
Protocols/Standards SIP RFC3261,
TCP/IP/UDP,
RTP,
HTTP/HTTPS,
ARP/RARP,
ICMP,
DNS (A record,
SRV,
NAPTR),
DHCP,
PPPoE,
TELNET,
TFTP,
NTP,
STUN,
TR-069, 802.1x
Networking Interfaces Dual-switched 10/100/1000Mbps port PoE
Feature Keys 3 SIP account,
2 line keys,
3-way conferencing,
3 XML programmable context-sensitive soft keys
Voice Codec Support for G.723.1,
G.729A/B,
G.711µ/a,
G.726-32,
G.722 (wide-band),
iLBC,
in-band and out-of-band DTMF (in audio,
RFC2833,
SIP INFO
Telephony Features Hold,
transfer,
forward,
3-way conference (via programmable key),
call waiting,
off-hook auto dial,
click-to-dial,
flexible dial plan,
personalized music ringtones,
server redundancy and fail-over
HD Audio Yes,
HD handset with support for wideband audio
Base Stand Yes,
1 angle position available
GRANDSTREAM
VoIP phone (Grandstream GXP1630)
* For special orders only
Net Price: 78.20 USD  Unit: pcs  VAT: 23%

#08840

VoIP phone (Grandstream GXP2130 v2)

The Linux-based GXP2130 is a standard enterprise-grade IP phone that features up to 3 lines, 4 XML programmable soft keys, 8 programmable BLF extension keys, dual Gigabit network ports, and 4-way voice conferencing. A 2.8 inch color LCD screen and HD audio allow for a crisp display and high quality calls. The GXP2130 comes equipped with Electronic Hook Switch (EHS) support for Plantronics headsets to allow for flexibility. The phone also comes pre-loaded with weather and currency exchange apps. Ideal for SMBs, enterprises and SOHOs, the GXP2130 is the perfect choice for users looking for a high quality, feature rich IP phone with advanced functionality that is simple to use.
SIP Compliant and Protocols SIP RFC3261,
TCP/IP/UDP,
RTP/RTCP,
HTTP/HTTPS,
ARP,
ICMP,
DNS (rekord A,
SRV,
NAPTR),
DHCP,
PPPoE,
TELNET,
TFTP,
NTP,
STUN,
SIMPLE,
LLDP,
LDAP,
TR-069, 802.1x,
TLS,
SRTP,
IPv6
Networking Interfaces Dual 10/100/1000mbps Ethernet ports,
PoE,
Bluetooth
Voice Codecs G.729A/B,
G.711µ/a-law,
G.726,
G.722 (szerokopasmowy),
i iLBC,
in-band and out-of-band DTMF (in audio,
RFC2833,
SIP INFO)
Superb Audio Quality Advanced Digital Signal Processing (DSP),
Silence suppression,
VAD,
CNG,
AGC high fidelity wideband audio (G.722). HD handset
Custom Ringtone Software Convert most music files to a Grandstream ringtone
Advanced Functionality Multi-line support,
multi-party conferencing (5-way),
multi-language support (MLS),
headset enabled,
expandable,
intercom,
AES encryption,
etc.
Cost-effective IP Solution Small to Medium enterprise office IP Phone
Warranty 24 months
GRANDSTREAM
VoIP phone (Grandstream GXP2130 v2)
* For special orders only
Net Price: 113.00 USD  Unit: pcs  VAT: 23%

#06006

VoIP phone (Grandstream GXP2135)

The GXP2135 is the ideal selection for busy users who value call control, productivity and usability, and manage medium to heavy call volumes. Equipped with 8 lines and 4 SIP accounts, a 2.8 inch color LCD display, and 32 digital speed dial/BLF keys, the GXP2135 enables quick and powerful usability.

As all Grandstream IP phones do, the GXP2135 features state-of-the-art security encryption technology (SRTP and TLS). The GXP2135 supports a variety of automated provisioning options, including zero-configuration with Grandstream’s UCM series IP PBXs, encrypted XML files and TR-069, to make mass deployment extremely easy.
SIP Compliant and Protocols SIP RFC3261,
TCP/IP/UDP,
RTP/RTCP,
HTTP/HTTPS,
ARP,
ICMP,
DNS (rekord A,
SRV,
NAPTR),
DHCP,
PPPoE,
TELNET,
TFTP,
NTP,
STUN,
SIMPLE,
LLDP,
LDAP,
TR-069, 802.1x,
TLS,
SRTP,
IPv6
Networking Interfaces Dual 10/100/1000mbps Ethernet ports,
PoE,
Bluetooth
Voice Codecs G.729A/B,
G.711µ/a-law,
G.726,
G.722 (szerokopasmowy),
i iLBC,
in-band and out-of-band DTMF (in audio,
RFC2833,
SIP INFO)
Superb Audio Quality Advanced Digital Signal Processing (DSP),
Silence suppression,
VAD,
CNG,
AGC high fidelity wideband audio (G.722). HD handset
Custom Ringtone Software Convert most music files to a Grandstream ringtone
Advanced Functionality Multi-line support,
multi-party conferencing (5-way),
multi-language support (MLS),
headset enabled,
expandable,
intercom,
AES encryption,
etc.
Cost-effective IP Solution Small to Medium enterprise office IP Phone
Warranty 24 months
GRANDSTREAM
VoIP phone (Grandstream GXP2135)
* For special orders only
Net Price: 113.00 USD  Unit: pcs  VAT: 23%

#08839

VoIP phone (Grandstream GXP2140)

A versatile Enterprise IP phone, the GXP2140 is a Linux-based device that includes 4 lines, 5 XML programmable soft keys, and 5-way conferencing. A 4.3 inch color LCD screen and HD audio allow for a crisp display and high quality calls. The GXP2140 comes equipped with Bluetooth, USB and EHS capabilities for flexibility. The phone also comes pre-loaded with weather & currency exchange apps. Add up to 4 GXP2200EXT modules to view an additional 160 lines, and customize your language for global use.
SIP Compliant and Protocols SIP RFC3261,
TCP/IP/UDP,
RTP/RTCP,
HTTP/HTTPS,
ARP,
ICMP,
DNS (rekord A,
SRV,
NAPTR),
DHCP,
PPPoE,
TELNET,
TFTP,
NTP,
STUN,
SIMPLE,
LLDP,
LDAP,
TR-069, 802.1x,
TLS,
SRTP,
IPv6
Networking Interfaces Dual 10/100/1000mbps Ethernet ports,
PoE,
Bluetooth
Voice Codecs G.729A/B,
G.711µ/a-law,
G.726,
G.722 (szerokopasmowy),
i iLBC,
in-band and out-of-band DTMF (in audio,
RFC2833,
SIP INFO)
Superb Audio Quality Advanced Digital Signal Processing (DSP),
Silence suppression,
VAD,
CNG,
AGC high fidelity wideband audio (G.722). HD handset
Custom Ringtone Software Convert most music files to a Grandstream ringtone
Advanced Functionality Multi-line support,
multi-party conferencing (5-way),
multi-language support (MLS),
headset enabled,
expandable,
intercom,
AES encryption,
etc.
Cost-effective IP Solution Small to Medium enterprise office IP Phone
Warranty 24 months
GRANDSTREAM
VoIP phone (Grandstream GXP2140)
* For special orders only
Net Price: 129.00 USD  Unit: pcs  VAT: 23%

#08841

VoIP phone (Grandstream GXP2160)

Our most powerful Enterprise IP Phone, the GXP2160 is a Linux-based device with 6 lines, 5 XML programmable soft keys, and 5-way conferencing. HD audio and a 4.3" color LCD screen create high quality calls, while the 24 BLF keys, Bluetooth, USB and EHS add versatility. The GXP2160 is perfect for Enterprise & SMB customers with the need for quality and versatility in their desktop communications.
SIP Compliant and Protocols 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
Networking Interfaces Dual 10/100/1000mbps Ethernet ports,
PoE,
Bluetooth
Voice Codecs G.729A/B,
G.711µ/a-law,
G.726,
G.722 (szerokopasmowe),
i iLBC,
in-band i out-of-band DTMF (in audio,
RFC2833,
SIP INFO)
Superb Audio Quality Advanced Digital Signal Processing (DSP),
Silence suppression,
VAD,
CNG,
AGC high fidelity wideband audio (G.722). HD handset
Custom Ringtone Software Convert most music files to a Grandstream ringtone
Advanced Functionality Multi-line support,
multi-party conferencing (5-way),
multi-language support (MLS),
headset enabled,
expandable,
intercom,
AES encryption,
etc.
Cost-effective IP Solution Small to Medium enterprise office IP Phone
Warranty 24 months
GRANDSTREAM
VoIP phone (Grandstream GXP2160)
* For special orders only
Net Price: 149.00 USD  Unit: pcs  VAT: 23%

#06007

VoIP phone (Grandstream GXP2170)

The GXP2170 is a powerful High-End IP phone that is ideal for busy users who handle high call volumes. Receptionists, administrators, sales staff and other call-intensive rolls can enjoy efficiency by utilizing the GXP2170’s 12 line keys, 4.3 inch color display LCD and 48 digital, on-screen speed dial/BLF keys. Provide users with the fastest possible connection speeds thanks to the device’s dual Gigabit, PoE network ports. Maximized call control, expandable speed dial/BLF capabilities and a sleek design makes this phone the ultimate high-volume experience.

As all Grandstream IP phones do, the GXP2170 features state-of-the-art security encryption technology (SRTP and TLS). The GXP2170 supports a variety of automated provisioning options, including zero-configuration with Grandstream’s UCM series IP PBXs, encrypted XML files and TR-069, to make mass deployment extremely easy.
SIP Compliant and Protocols 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
Networking Interfaces Dual 10/100/1000mbps Ethernet ports,
PoE,
Bluetooth
Voice Codecs G.729A/B,
G.711µ/a-law,
G.726,
G.722 (szerokopasmowe),
i iLBC,
in-band i out-of-band DTMF (in audio,
RFC2833,
SIP INFO)
Superb Audio Quality Advanced Digital Signal Processing (DSP),
Silence suppression,
VAD,
CNG,
AGC high fidelity wideband audio (G.722). HD handset
Custom Ringtone Software Convert most music files to a Grandstream ringtone
Advanced Functionality Multi-line support,
multi-party conferencing (5-way),
multi-language support (MLS),
headset enabled,
expandable,
intercom,
AES encryption,
etc.
Cost-effective IP Solution Small to Medium enterprise office IP Phone
Warranty 24 months
GRANDSTREAM
VoIP phone (Grandstream GXP2170)
* For special orders only
Net Price: 149.00 USD  Unit: pcs  VAT: 23%