Produkty a služby
Grandstream GXP 1100
GXP 1100/1105 je ďalšia generácia IP telefónov pre malé a stredné spoločnosti s jedným SIP kontom, 4 programovatelé tlačitka a jedno sieťové pripojenie. GXP 1100 poskytuje zvuk v HD kvalite, automatický a jednoduchy provisioning pre jednoduché nasadenie, rozšírenú ochranu a bezpečnosť. Tento telefón je ideálny pre menšie firmy, do hotelov a lobby pre vysokú kvalitu zvuku a atraktívnu cenu.
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Základné vlastnosti
-Single SIP account and up to 2 calls, 4 programmable keys
-HD handset with support for wideband audio
-Single 10/100Mbps network port
-7 dedicated function keys for Hold, Flash/Call-Waiting, Transfer, Message, Mute, Volume, Dial/Redial
-Automated provisioning using TR-069 or AES encrypted XML configuration file, SRTP and TLS for advanced security protection, 802.1x for media access control
Špecifikácia
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: Single 10/100Mbps port
Feature Keys: 1 SIP account, 4 programmable keys, 7 dedicated function keys for HOLD, FLASH, TRANSFER, MUTE, VOLUME, REDIAL and MESSAGE (with LED indicator)
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, hot-desking, 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 GXP 1105
GXP 1100/1105 je ďalšia generácia IP telefónov pre malé a stredné spoločnosti s jedným SIP kontom, 4 programovatelé tlačitka, jedno sieťové pripojenie a integrované PoE. GXP 1100 poskytuje zvuk v HD kvalite, automatický a jednoduchy provisioning pre jednoduché nasadenie, rozšírenú ochranu a bezpečnosť. Tento telefón je ideálny pre menšie firmy, do hotelov a lobby pre vysokú kvalitu zvuku a atraktívnu cenu.
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Základné vlastnosti
-Single SIP account and up to 2 calls, 4 programmable keys
-HD handset with support for wideband audio
-Single 10/100Mbps network port
-7 dedicated function keys for Hold, Flash/Call-Waiting, Transfer, Message, Mute, Volume, Dial/Redial
-Automated provisioning using TR-069 or AES encrypted XML configuration file, SRTP and TLS for advanced security protection, 802.1x for media access control
Špecifikácia
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: Single 10/100Mbps port, integrated PoE
Feature Keys: 1 SIP account, 4 programmable keys, 7 dedicated function keys for HOLD, FLASH, TRANSFER, MUTE, VOLUME, REDIAL and MESSAGE (with LED indicator)
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, hot-desking, 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 GXP 280
Grandstream GXP280 je nákladovo efektívnou voľbou pre SOHO užívateľov, ktorí potrebujú možnosť širokého výberu funkcií pri cenovej dostupnosti 1-linkového SIP telefónu.Grandstream GXP280 prináša vynikajúcu kvalitu zvuku, komplexné telefónne funkcie, automatické zabezpečenie bezpečnostnej ochrany súkromia a širokú interoperabilitu s väčšinou SIP zariadeniami tretích strán, bránami a soft-switch platformami.
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Základné vlastnosti
• Jednoduché línie vzhľadu, s funkciou podržania hovoru umožňuje
• spracovať 2 simultánne hovory
• Duálne 10/100Mbps switch Ethernet porty
• 128 x 32 grafické LCD
• Podpora viacerých jazykov (angličtina, španielčina, nemčina, francúzština a ďalšie)
• 3 XML programovateľné kontextové soft keys
• podpora náhlavnej súpravy (2,5 mm a RJ-9 jack)
• Plne duplexný hlasitý telefón s pokročilým potlačením akustických ozvien
• High fidelity širokopásmový zvuk
• SRTP
• Možnosť pripevnenia na stenu
Špecifikácia
Protocols/Standards: SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP, ARP/RARP, ICMP, DNS (A record and SRV), DHCP, PPPoE, TFTP, NTP, STUN, SIMPLE
Network Interfaces: Dual switched 10/100 Mbps
Graphic Display: 128 x 32 pixel graphic LCD
Feature Keys: 3 dynamic context sensitive soft keys, 3 navigation/menu/volume keys, 7 dedicated function keys for: HOLD, SPEAKERPHONE, SEND, TRANSFER, CONFERENCE, FLASH and HEADSET
Voice Codec: Support for G.723.1, G.729 A/B, G.711 u/a-law, G.726, G.722 (wide-band), GSM and iLBC, in- band and out-of-band DTMF (in audio, RFC2833, SIP INFO)
Telephony Features: Hold, Mute, Transfer, Forward, 3-way Conference, Downloadable Phone Book (XML, LDAP up to 200 items), Call Waiting, Call Log, Off-hook Auto Dial, Auto Answer, Click-To-Dial, Downloadable Ringtones, Server Redundancy and Fail-over Support
Headset Jack: 2.5mm and RJ9 headset jack
QoS: Layer 2 (802.1Q, 801.2p) and Layer 3 (ToS, DiffServ, MPLS) QoS
Security: User and administrator level passwords, MD5 and MD5- sess based authentication, AES based secure configuration file, SRTP
Multi-language English, Spanish, French, German, etc
Upgrade/Provisioning: Firmware upgrade via TFTP/HTTP, mass deployment using central secure provisioning file
Power: Universal power adaptor (Input: 100-240VAC 50-60Hz, Output:+5VDC, 1.2A, UL Certified) included
Physical: Dimension: 168mmx200mmx89.5mm Unit weight: 0.62KG Package weight: 0.725KG
Temperature/Humidity: 32-104°F/ 0-40 °C, 10-90% (non-condensing)
Compliance: FCC,CE,C-Tick
Grandstream GXP 285
Grandstream GXP285 je nákladovo efektívnou voľbou pre SOHO užívateľov, ktorí potrebujú možnosť širokého výberu funkcií pri cenovej dostupnosti 1-linkového SIP telefónu napájaním cez PoE. Grandstream GXP285 prináša vynikajúcu kvalitu zvuku, komplexné telefónne funkcie, automatické zabezpečenie bezpečnostnej ochrany súkromia a širokú interoperabilitu s väčšinou SIP zariadeniami tretích strán, bránami a soft-switch platformami.
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Základné vlastnosti
• dvojriadkový 128 x 32 grafický LCD displej
• všetky štandardné telefónne funkcie (s funkciou podržania hovoru umožňuje spracovať 2 simultánne hovory)
• Integrované PoE
• duálne 10/100Mbps switch Ethernet porty
• podpora náhlavnej súpravy (2,5 mm a RJ-9 jack)
• 1 SIP účet
• 3 funkčné klávesy
• podpora 3 - strannej konferencie
• podpora hlasitého hovoru
• jednoduchá obsluha
• SRTP
• podpora viacerých jazykov vrátane slovenčiny
• možnosť pripevnenia na stenu
Špecifikácia
Protocols/Standards: SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP, ARP/RARP, ICMP, DNS (A record and SRV), DHCP, PPPoE, TFTP, NTP, STUN, SIMPLE
Network Interfaces: Dual switched 10/100 Mbps with integrated PoE support
Graphic Display: 128 x 32 pixel graphic LCD
Feature Keys: 3 dynamic context sensitive soft keys, 3 navigation/menu/volume keys, 7 dedicated function keys for: HOLD, SPEAKERPHONE, SEND, TRANSFER, CONFERENCE, FLASH and HEADSET
Voice Codec: Support for G.723.1, G.729 A/B, G.711 u/a-law, G.726, G.722 (wide-band), GSM and iLBC, in- band and out-of-band DTMF (in audio, RFC2833, SIP INFO)
Telephony Features: Hold, Mute, Transfer, Forward, 3-way Conference, Downloadable Phone Book (XML, LDAP up to 200 items), Call Waiting, Call Log, Off-hook Auto Dial, Auto Answer, Click-To-Dial, Downloadable Ringtones, Server Redundancy and Fail-over Support
Headset Jack: 2.5mm and RJ9 headset jack
QoS: Layer 2 (802.1Q, 801.2p) and Layer 3 (ToS, DiffServ, MPLS) QoS
Security: User and administrator level passwords, MD5 and MD5- sess based authentication, AES based secure configuration file, SRTP
Multi-language English, Spanish, French, German, etc
Upgrade/Provisioning: Firmware upgrade via TFTP/HTTP, mass deployment using central secure provisioning file
Power: Universal power adaptor (Input: 100-240VAC 50-60Hz, Output:+5VDC, 1.2A, UL Certified) included
Physical: Dimension: 168mmx200mmx89.5mm Unit weight: 0.62KG Package weight: 0.725KG
Temperature/Humidity: 32-104°F/ 0-40 °C, 10-90% (non-condensing)
Compliance: FCC,CE,C-Tick
Grandstream GXP 1400
GXP1400 is a next generation small-to-medium business IP phone that features 2 lines with 2 SIP account, a 128×40 graphical LCD, 3 XML programmable context-sensitive soft keys, dual network ports, and 3-way conference. The GXP1400 delivers superior HD audio quality, rich and leading edge telephony features, personalized information and customizable application service, automated provisioning for easy deployment, advanced security protection for privacy, and broad interoperability with most 3rd party SIP devices and leading SIP/NGN/IMS platforms. It is a perfect choice for small-to-medium businesses looking for a high quality, feature rich IP phone with affordable cost.
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Základné vlastnosti
- 2 linkové tlačidlá s dvojfarebnými LED diódami (1 SIP účet s obsluhou 2 liniek)
- bez PoE, napájanie cez adaptér, ktorý je súčasťou balenia
- grafický LCD displej 128x40
- 3 XML programovateľné soft tlačidlá
- konferečný hovor do 3 účastníkov
- HD kvalita zvuku (slúchadlo, reproduktor hlasitého odposluchu) s pokročilým potlačením akustických ozvien
- automatizovaná osobná informačná služba (napr. lokálna predpoveď počasia, kurzy mien, burzové indexy)
- personalizované hudobné tóny zvonenia
- podpora downloadu telefónneho zoznamu (XML, LDAP) do 500 záznamov
- zoznam hovorov – posledných 200 záznamov
Špecifikácia
SIP Compliant and Protocols: SIP RFC3261, TCP/IP/UDP, RTP, HTTP/HTTPS, ARP/RARP, ICMP, DNS (A record, SRV, NAPTR), DHCP, PPPoE, TELNET, TFTP, NTP, STUN, SIMPLE, TR-069, 802.1x.
Networking Interfaces: Dual switched 10/100Mbps ports
Graphical Display: 128x40 graphical LCD display
Feature Keys: 2 line keys with dual-color LED and 2 SIP account, 3 XML programmable context sensitive soft keys, 5 navigation/menu/volume keys, 8 dedicated function keys for HOLD, TRANSFER, CONFERENCE, VOLUME, HEADSET, MUTE, SPEAKERPHONE, SEND/REDIAL
Voice Codecs: 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, call park, pickup, shared-call-appearance (SCA)/bridged-line-appearance (BLA), downloadable phone book (XML, LDAP, up to 500 items), call waiting, call log (up to 200 records), off-hook auto dial, auto answer, click-to-dial, flexible dial plan, hot desking, personalized music ringtones, server redundancy and fail-over
HD Audio: Yes, HD handset with support for wideband audio
Headset Jack: RJ9 headset jack
Base Stand: Yes, 2 angle positions available
Wall Mountable: Yes
QoS: Layer 2 (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS
Security: User and administrator level passwords, MD5 and MD5-sess based authentication, AES encrypted configuration file, SRTP, TLS, 802.1x media access control
Multi-language: Slovakia, English, German, Italian, French, Spanish, Portuguese, Russian, Croatian, simplified and traditional Chinese, Korean, Japanese and etc
Upgrade/Provisioning: Firmware upgrade via TFTP/HTTP/HTTPS, mass provisioning using TR-069 or AES encrypted XML configuration file
Power & Green Energy Efficiency: Universal power adapter included: Input: 100-240VAC 50-60Hz; Output: +5VDC, 800mA Max power consumption 2.5W (power adapter)
Physical: Unit dimension: 186mm (W) x 210mm (L) x 81mm (D) Unit
weight: 0.7KG Package weight: 1.1KG (GXP1400)
Operating Temperature & Humidity: 32-104oF/0-40oC, 10-90% (non-condensing)
Package Content: GXP1400 phone, handset with cord, base stand, universal power supply, network cable, quick start guide
Compliance: FCC Part 15 (CFR 47) Class B EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3, EN 60950-1 AS/NZS CISPR 22 Class B, AS/NZS CISPR 24, RoHS UL 60950 (power adapter)
Grandstream GXP 1405
GXP1405 is a next generation small-to-medium business IP phone that features 2 lines with 2 SIP account, a 128×40 graphical LCD, 3 XML programmable context-sensitive soft keys, dual network ports with integrated PoE, and 3-way conference. The GXP1405 delivers superior HD audio quality, rich and leading edge telephony features, personalized information and customizable application service, automated provisioning for easy deployment, advanced security protection for privacy, and broad interoperability with most 3rd party SIP devices and leading SIP/NGN/IMS platforms. It is a perfect choice for small-to-medium businesses looking for a high quality, feature rich IP phone with affordable cost.
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Základné vlastnosti
- 2 linkové tlačidlá s dvojfarebnými LED diódami (1 SIP účet s obsluhou 2 liniek)
- integrované PoE, bez adaptéru
- grafický LCD displej 128x40
- 3 XML programovateľné soft tlačidlá
- konferečný hovor do 3 účastníkov
- HD kvalita zvuku (slúchadlo, reproduktor hlasitého odposluchu) s pokročilým potlačením akustických ozvien
- automatizovaná osobná informačná služba (napr. lokálna predpoveď počasia, kurzy mien, burzové indexy)
- personalizované hudobné tóny zvonenia
- podpora downloadu telefónneho zoznamu (XML, LDAP) do 500 záznamov
- zoznam hovorov – posledných 200 záznamov
Špecifikácia
SIP Compliant and Protocols: SIP RFC3261, TCP/IP/UDP, RTP, HTTP/HTTPS, ARP/RARP, ICMP, DNS (A record, SRV, NAPTR), DHCP, PPPoE, TELNET, TFTP, NTP, STUN, SIMPLE, TR-069, 802.1x.
Networking Interfaces: Dual switched 10/100Mbps ports with integrated PoE
Graphical Display: 128x40 graphical LCD display
Feature Keys: 2 line keys with dual-color LED and 2 SIP account, 3 XML programmable context sensitive soft keys, 5 navigation/menu/volume keys, 8 dedicated function keys for HOLD, TRANSFER, CONFERENCE, VOLUME, HEADSET, MUTE, SPEAKERPHONE, SEND/REDIAL
Voice Codecs: 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, call park, pickup, shared-call-appearance (SCA)/bridged-line-appearance (BLA), downloadable phone book (XML, LDAP, up to 500 items), call waiting, call log (up to 200 records), off-hook auto dial, auto answer, click-to-dial, flexible dial plan, hot desking, personalized music ringtones, server redundancy and fail-over
HD Audio: Yes, HD handset with support for wideband audio
Headset Jack: RJ9 headset jack
Base Stand: Yes, 2 angle positions available
Wall Mountable: Yes
QoS: Layer 2 (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS
Security: User and administrator level passwords, MD5 and MD5-sess based authentication, AES encrypted configuration file, SRTP, TLS, 802.1x media access control
Multi-language: Slovakia, English, German, Italian, French, Spanish, Portuguese, Russian, Croatian, simplified and traditional Chinese, Korean, Japanese and etc
Upgrade/Provisioning: Firmware upgrade via TFTP/HTTP/HTTPS, mass provisioning using TR-069 or AES encrypted XML configuration file
Power & Green Energy Efficiency: Universal power adapter included: Input: 100-240VAC 50-60Hz; Output: +5VDC, 800mA Integrated PoE (802.3af) Max power consumption 3W (PoE)
Physical: Unit dimension: 186mm (W) x 210mm (L) x 81mm (D) Unit
weight: 0.7KG Package weight: 1.0KG (GXP1405)
Operating Temperature & Humidity: 32-104oF/0-40oC, 10-90% (non-condensing)
Package Content: GXP1405 phone, handset with cord, base stand, network cable, quick start guide
Compliance: FCC Part 15 (CFR 47) Class B EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3, EN 60950-1 AS/NZS CISPR 22 Class B, AS/NZS CISPR 24, RoHS UL 60950 (power adapter)
Grandstream GXP 1450 HD Enterprise IP Phone
GXP1450 is a next generation enterprise grade IP phone that features 2 lines with 2 SIP accounts, a 180×60 backlit graphical LCD, 3 XML programmable context-sensitive soft keys, dual network ports with integrated PoE, and 3-way conference. The GXP1450 delivers superior HD audio quality, rich and leading edge telephony features, personalized information and customizable application service, automated provisioning for easy deployment, advanced security protection for privacy, and broad interoperability with most 3rd party SIP devices and leading SIP/NBN/IMS platforms. It is a perfect choice for enterprise users looking for a high quality, feature rich IP phone with affordable cost.
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Základné vlastnosti
- 2 SIP účty
- Podsvietený grafický LCD displej 180 x 60 so 4 stupňami nastavenia úrovne šedej farby
- 3 programovateľné tlačidlá
- 3-cestná konferencia
- HD audio
- Full duplex reproduktor
- Podpora viacerých jazykov vrátane slovenčiny
- XML screen
- Personalizované webové aplikácie
Špecifikácia
SIP Compliant and Protocols: SIP RFC2361, TCP/IP/UDP, RTP, HTTP/HTTPS, ARP/RARP, ICMP, DNS (A record, SRV, NAPTR), DHCP, PPPoE, TELNET, TFTP, NTP, STUN, SIMPLE, TR-069, 802.1x etc.
Networking Interfaces: Dual switched 10/100Mbps ports with integrated PoE
Graphical Display: Back-lit 180x60 graphical LCD display with up to 4 level grayscale
Feature Keys: 2 line keys with dual-color LED and 2 independent SIP accounts, 3 XML programmable context sensitive soft keys, 5 navigation/menu/volume keys, 10 dedicated function keys for: HOLD, SPEAKERPHONE, SEND/REDIAL, TRANSFER, CONFERENCE, MUTE, HEADSET, VOLUME, PHONEBOOK/CONTACTS, and MESSAGE (with LED indicator)
Voice Codecs: Support for G.723.1, G.729A/B, G.711u/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, call park, pickup, shared-call-appearance (SCA)/bridged-line-appearance (BLA), downloadable phone book (XML, LDAP, up to 500 items), call waiting, call log (up to 500 records), XML customization of screen, off-hook auto dial, auto answer, click-to-dial, flexible dial plan, hot desking, personalized music ringtones and music on hold, server redundancy and fail-over
HD Audio: Yes, both on handset and speakerphone
Headset Jack: RJ9 and 2.5mm headset jack
Base Stand: Yes, allow 2 angle positions
Wall Mountable: Yes
QoS: Layer 2 (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS
Security: User and administrator level passwords, MD5 and MD5-sess based authentication, AES based secure configuration file, SRTP, TLS, 802.1x media access control
Multi-language: Slovakia, English, German, Italian, French, Spanish, Portuguese, Russian, Croatian, simplified and traditional Chinese, Korean, Japanese, etc
Upgrade/Provisioning: Firmware upgrade via TFTP/HTTP/HTTPS, mass provisioning using TR-069 or encrypted XML configuration file
Power & Green Energy Efficiency: Universal power adapter included: Input: 100-240VAC 50-60Hz; Output: +5VDC, 800mA; Integrated Power-over-Ethernet (802.3af); Max power consumption 2.5W (power adapter)
Physical: Unit dimension: 186mm (W) x 210mm (L) x 81mm (D); Unit weight: 0.8KG; Package weight: 1.3KG)
Operating Temperature & Humidity: 32-104 F / 0-40 C, 10-90% (non-condensing)
Package Content: GXP1450 phone, handset with cord, base stand, universal power supply, network cable, Quick Start Guide
Compliance: FCC Part 15 (CFR 47) Class B; EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3, EN 60950-1; AS/NZS CISPR 22 Class B, AS/NZS CISPR 24, RoHS; UL 60950 (power adapter)
Cisco SPA922 1-Line IP Phone with 2-Port Switch
Stylish and functional in design, the SPA-922 IP phone is ideal for a residence or business using a hosted IP telephony service, an IP PBX, or a large-scale IP Centrex deployment. The SPA922 leverages industry leading VoIP technology from Linksys to deliver an upgradeable high quality IP phone that is unparalleled in features, value, and support.
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Standard features on the SPA-922 include dual switched Ethernet ports, 802.3af PoE, a high resolution graphical display, speakerphone and a 2.5 mm head-set port. The SPA922 supports one line with two call appearances and provides support for three way conferencing, attended call transfer, and placing a call on hold to answer an incoming call. The line can be configured as a unique phone number (or extension), or can be configured to share a number that is assigned to multiple phones.
Features:
Single Line
802.3af PoE Compliant
Two Port Ethernet Switch
Protocols: SIP
- Full featured one-line business class IP phone supporting Power over Ethernet 802.3af
- Connect directly to an Internet Telephone Service Provider or connect to an IP PBX
- Dual switched Ethernet ports, Speakerphone, Caller ID, Call Hold, Conferencing, and more
- Easy installation with secure remote provisioning. Menu based and web based configuration.
Data Sheet
Cisco SPA922 VoIP Phone
Links
Cisco SPA922